
From mahesh.awaradi@globaledgesoft.com  Tue Dec  1 21:14:22 2009
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Subject: [Sip] 182 call queued
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Hi All ,
Can any body tell me  182 call queued packet format.

Regards
Mahesh



From abhatnagar192006@gmail.com  Tue Dec  1 23:28:38 2009
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Date: Wed, 2 Dec 2009 07:58:21 +0030
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From: aayush bhatnagar <abhatnagar192006@gmail.com>
To: Mahesh <mahesh.awaradi@globaledgesoft.com>
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Subject: Re: [Sip] 182 call queued
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What do you mean when you say 'packet format' ? The message structure ?

On 02/12/2009, Mahesh <mahesh.awaradi@globaledgesoft.com> wrote:
> Hi All ,
> Can any body tell me  182 call queued packet format.
>
> Regards
> Mahesh
>
>
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>


-- 
aayush
http://in.linkedin.com/in/abhatnagar19
======

From dworley@nortel.com  Wed Dec  2 14:27:27 2009
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On Wed, 2009-12-02 at 10:44 +0530, Mahesh wrote:
> Can any body tell me  182 call queued packet format.

It is an ordinary SIP provisional response with response code 182.
There is no special format.

Dale



From spencer@wonderhamster.org  Tue Dec  8 14:25:52 2009
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From: "Spencer Dawkins" <spencer@wonderhamster.org>
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Cc: techwg@sipforum.org
Subject: [Sip] Fw: [martini] New Mailing List announcement - MARTINI (was - DREGS)
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Just so everyone is on the same page, Cullen asked me to announce this 
mailing list on SIP, SIPPING, and the techwg mailing list...

See you on MARTINI!

Thanks,

Spencer

----- Original Message ----- 
From: "Spencer Dawkins" <spencer@wonderhamster.org>
To: <dispatch@ietf.org>; <rai@ietf.org>
Cc: <martini@ietf.org>
Sent: Tuesday, December 08, 2009 4:23 PM
Subject: [martini] New Mailing List announcement - MARTINI (was - DREGS)


> Cullen has asked me to announce a new mailing list, called MARTINI 
> (acronym
> for "Multiple AoR reachabiliTy InformatioN Indication"), to be used for
> discussion of what has variously been called "implicit registration",
> "domain registration", and probably some other terms as well.
>
> To be a little more precise, the MARTINI mailing list was created to 
> discuss
> means by which an entity that is authoritative for SIP URIs can 
> dynamically
> register reachability information for multiple Addresses of Record 
> ("AORs")
> with a service provider.
>
> The coordinates are in the place you'd expect:
>
> General Discussion: martini@ietf.org
> To Subscribe: martini-request@ietf.org
> In Body: subscribe
> Archive: http://www.ietf.org/mail-archive/web/martini/index.html
>
> Just to catch people up... DISPATCH recommended that this proposed work go
> forward in a mini-working group. The IESG approved a proposed charter to 
> be
> sent for a two-week review on last Thursday's telechat, so you should see
> that announcement popping out fairly quickly.
>
> That charter also proposes an interim meeting in January 2010, and we'll 
> be
> discussing logistics on the MARTINI mailing list in the near future, so
> please subscribe Real Soon Now if you are interested in the topic.
>
> Thanks!
>
> Spencer
> _______________________________________________
> martini mailing list
> martini@ietf.org
> https://www.ietf.org/mailman/listinfo/martini 


From nitinkapoorr@gmail.com  Thu Dec 10 05:11:00 2009
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From: Nitin Kapoor <nitinkapoorr@gmail.com>
To: sip@ietf.org
Content-Type: multipart/alternative; boundary=00032557b162fe8d16047a5f8864
Subject: [Sip] Two Media Stream in One call
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--00032557b162fe8d16047a5f8864
Content-Type: text/plain; charset=ISO-8859-1

Hello All,

I am facing the issue with one of my customer where he is getting two media
streams in one call and they are using SBC(Nextone)

I checked the signaling traces and everything looks okay to me as well as
the RTP traces too.

As per the traces one  RTP packet is going from 64.156.174.71 and the other
is coming from 8.15.244.11.

Could you please let me know is it possible where i can have the two media
streams in one call when this is the not the conference call. And if this is
possible then what is the scenario where its possible.

Please find the attached traces for the same.

Thanks,
Nitin Kapoor
General Telecom

--00032557b162fe8d16047a5f8864
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

Hello All,<br>
<div class=3D"gmail_quote"><br>I am facing the issue with one of my custome=
r where he is getting two media streams in one call and they are using SBC(=
Nextone)<br><br>I checked the signaling traces and everything looks okay to=
 me as well as the RTP traces too.<br>

<br>As per the traces one=A0 RTP packet is going from 64.156.174.71 and the=
 other is coming from 8.15.244.11. <br><br>Could
you please let me know is it possible where i can have the two media
streams in one call when this is the not the conference call. And if
this is possible then what is the scenario where its possible.<br>
<br>Please find the attached traces for the same.<br><br>Thanks,<br><font c=
olor=3D"#888888">Nitin Kapoor<br>General Telecom<br>
</font></div>

--00032557b162fe8d16047a5f8864--

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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Nitin Kapoor'" <nitinkapoorr@gmail.com>, <sip@ietf.org>
References: <c80c92d0912100510l3933eb4cud5c76c78413499c7@mail.gmail.com>
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Date: Thu, 10 Dec 2009 15:32:38 +0200
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Subject: Re: [Sip] Two Media Stream in One call
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Hi,

To know the answer you need to provide the SDP in the offer and answer.

If the offer/answer SDP defined two media streams, or N media streams this
is probably OK.

Roni Even

 

From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of Nitin
Kapoor
Sent: Thursday, December 10, 2009 3:11 PM
To: sip@ietf.org
Subject: [Sip] Two Media Stream in One call

 

Hello All,


I am facing the issue with one of my customer where he is getting two media
streams in one call and they are using SBC(Nextone)

I checked the signaling traces and everything looks okay to me as well as
the RTP traces too.

As per the traces one  RTP packet is going from 64.156.174.71 and the other
is coming from 8.15.244.11. 

Could you please let me know is it possible where i can have the two media
streams in one call when this is the not the conference call. And if this is
possible then what is the scenario where its possible.

Please find the attached traces for the same.

Thanks,
Nitin Kapoor
General Telecom


------=_NextPart_000_004C_01CA79AE.06A07590
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<div class=3DSection1>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Hi,<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>To know the answer you need to provide the SDP in the =
offer and
answer.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>If the offer/answer SDP defined two media streams, or N =
media
streams this is probably OK.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Roni Even<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<div style=3D'border:none;border-left:solid blue 1.5pt;padding:0in 0in =
0in 4.0pt'>

<div>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt =
0in 0in 0in'>

<p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] <b>On Behalf Of =
</b>Nitin
Kapoor<br>
<b>Sent:</b> Thursday, December 10, 2009 3:11 PM<br>
<b>To:</b> sip@ietf.org<br>
<b>Subject:</b> [Sip] Two Media Stream in One call<o:p></o:p></span></p>

</div>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Hello All,<o:p></o:p></p>

<div>

<p class=3DMsoNormal><br>
I am facing the issue with one of my customer where he is getting two =
media
streams in one call and they are using SBC(Nextone)<br>
<br>
I checked the signaling traces and everything looks okay to me as well =
as the
RTP traces too.<br>
<br>
As per the traces one&nbsp; RTP packet is going from 64.156.174.71 and =
the
other is coming from 8.15.244.11. <br>
<br>
Could you please let me know is it possible where i can have the two =
media
streams in one call when this is the not the conference call. And if =
this is
possible then what is the scenario where its possible.<br>
<br>
Please find the attached traces for the same.<br>
<br>
Thanks,<br>
<span style=3D'color:#888888'>Nitin Kapoor<br>
General Telecom</span><o:p></o:p></p>

</div>

</div>

</div>

</body>

</html>

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From christer.holmberg@ericsson.com  Thu Dec 10 05:56:25 2009
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From: "Christer Holmberg" <christer.holmberg@ericsson.com>
To: "Nitin Kapoor" <nitinkapoorr@gmail.com>, <sip@ietf.org>
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Hi,

It is possible.

In theory this could be done with a single c- line, because SDP only =
talks about where media is to be sent - not where it comes from.

In reality, however, I assume this is a case where each m- line has its =
own c- line, each with a different IP address.

Regards,

Christer


-----Original Message-----
From: sip-bounces@ietf.org on behalf of Nitin Kapoor
Sent: Thu 12/10/2009 2:10 PM
To: sip@ietf.org
Subject: [Sip] Two Media Stream in One call
=20
Hello All,

I am facing the issue with one of my customer where he is getting two =
media
streams in one call and they are using SBC(Nextone)

I checked the signaling traces and everything looks okay to me as well =
as
the RTP traces too.

As per the traces one  RTP packet is going from 64.156.174.71 and the =
other
is coming from 8.15.244.11.

Could you please let me know is it possible where i can have the two =
media
streams in one call when this is the not the conference call. And if =
this is
possible then what is the scenario where its possible.

Please find the attached traces for the same.

Thanks,
Nitin Kapoor
General Telecom


From nitinkapoorr@gmail.com  Thu Dec 10 07:47:13 2009
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--00235447067494a47e047a61b7ad
Content-Type: text/plain; charset=ISO-8859-1

> Hello,
>
> Below is the packet in UDP format, Could you please help me on this to
> understand this scenario. Also attached is the complete traces.
>
> INVITE sip:9996214018210532@208.85.248.42<sip%3A9996214018210532@208.85.248.42>SIP/2.0
>
> Max-Forwards: 69
>
> Session-Expires: 3600;refresher=uac
>
> Supported: timer, 100rel
>
> To: <sip:9996214018210532@208.85.248.42<sip%3A9996214018210532@208.85.248.42>
> >
>
> From: <sip:8003223223@tmsash1-15-sip>;tag=3468945569-250042
>
> Call-ID: 7044988-3468945569-250036@mscasheq01.soundbite.com
>
> CSeq: 1 INVITE
>
> Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER,
> SUBSCRIBE, PRACK, UPDATE
>
> Via: SIP/2.0/UDP 8.15.244.10:5060
> ;branch=z9hG4bKf5de4d0e3c0860d122139e1e30a2995c
>
> Contact: <sip:8003223223@8.15.244.10:5060>
>
> Call-Info:
> <sip:8.15.244.10>;method="NOTIFY;Event=telephone-event;Duration=1000"
>
> Content-Type: application/sdp
>
> Content-Length: 124
>
>
>
> v=0
>
> o=mscasheq01 547500705 547500705 IN IP4 8.15.244.10
>
> s=sip call
>
> c=IN IP4 8.15.244.11
>
> t=0 0
>
> m=audio 19612 RTP/AVP 0
>
> SIP/2.0 100 Trying
>
> Via: SIP/2.0/UDP 8.15.244.10:5060
> ;branch=z9hG4bKf5de4d0e3c0860d122139e1e30a2995c
>
> From: <sip:8003223223@tmsash1-15-sip>;tag=3468945569-250042
>
> To: <sip:9996214018210532@208.85.248.42<sip%3A9996214018210532@208.85.248.42>
> >
>
> Call-ID: 7044988-3468945569-250036@mscasheq01.soundbite.com
>
> CSeq: 1 INVITE
>
> Server: Sansay VSX 2.1
>
> Content-Length: 0
>
>
>
> SIP/2.0 180 Ringing
>
> Via: SIP/2.0/UDP 8.15.244.10:5060
> ;branch=z9hG4bKf5de4d0e3c0860d122139e1e30a2995c
>
> From: <sip:8003223223@tmsash1-15-sip>;tag=3468945569-250042
>
> To: <sip:9996214018210532@208.85.248.42<sip%3A9996214018210532@208.85.248.42>
> >;tag=228209687-tdb1145808424
>
> Call-ID: 7044988-3468945569-250036@mscasheq01.soundbite.com
>
> CSeq: 1 INVITE
>
> Server: Sansay VSX 2.1
>
> Content-Length: 0
>
>
>
> SIP/2.0 200 OK
>
> Via: SIP/2.0/UDP 8.15.244.10:5060
> ;branch=z9hG4bKf5de4d0e3c0860d122139e1e30a2995c
>
> From: <sip:8003223223@tmsash1-15-sip>;tag=3468945569-250042
>
> To: <sip:9996214018210532@208.85.248.42<sip%3A9996214018210532@208.85.248.42>
> >;tag=228209687-tdb1145808424
>
> Call-ID: 7044988-3468945569-250036@mscasheq01.soundbite.com
>
> CSeq: 1 INVITE
>
> Server: Sansay VSX 2.1
>
> Contact: <sip:9996214018210532@208.85.248.42:5060;transport=udp>
>
> Content-Type: application/sdp
>
> Content-Length:   158
>
>
>
> v=0
>
> o=sansay-VSX 10 10 IN IP4 64.156.174.71
>
> s=session controller
>
> c=IN IP4 64.156.174.71
>
> t=0 0
>
> m=audio 10766 RTP/AVP 0
>
> a=rtpmap:0 PCMU/8000
>
> a=ptime:20
>
> ACK sip:9996214018210532@208.85.248.42:5060;transport=udp SIP/2.0
>
> Max-Forwards: 69
>
> To: <sip:9996214018210532@208.85.248.42<sip%3A9996214018210532@208.85.248.42>
> >;tag=228209687-tdb1145808424
>
> From: <sip:8003223223@tmsash1-15-sip>;tag=3468945569-250042
>
> Call-ID: 7044988-3468945569-250036@mscasheq01.soundbite.com
>
> CSeq: 1 ACK
>
> Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER,
> SUBSCRIBE, PRACK, UPDATE
>
> Via: SIP/2.0/UDP 8.15.244.10:5060
> ;branch=z9hG4bK7208e1bfdd2962741dc3dee9d2576820
>
> Contact: <sip:8003223223@8.15.244.10:5060>
>
> Content-Length: 0
>
>
> Thanks,
> Nitin Kapoor
>
> 2009/12/10 Christer Holmberg <christer.holmberg@ericsson.com>
>
>
>> Hi,
>>
>> It is possible.
>>
>> In theory this could be done with a single c- line, because SDP only talks
>> about where media is to be sent - not where it comes from.
>>
>> In reality, however, I assume this is a case where each m- line has its
>> own c- line, each with a different IP address.
>>
>> Regards,
>>
>> Christer
>>
>>
>> -----Original Message-----
>> From: sip-bounces@ietf.org on behalf of Nitin Kapoor
>> Sent: Thu 12/10/2009 2:10 PM
>> To: sip@ietf.org
>> Subject: [Sip] Two Media Stream in One call
>>
>> Hello All,
>>
>> I am facing the issue with one of my customer where he is getting two
>> media
>> streams in one call and they are using SBC(Nextone)
>>
>> I checked the signaling traces and everything looks okay to me as well as
>> the RTP traces too.
>>
>> As per the traces one  RTP packet is going from 64.156.174.71 and the
>> other
>> is coming from 8.15.244.11.
>>
>> Could you please let me know is it possible where i can have the two media
>> streams in one call when this is the not the conference call. And if this
>> is
>> possible then what is the scenario where its possible.
>>
>> Please find the attached traces for the same.
>>
>> Thanks,
>> Nitin Kapoor
>> General Telecom
>>
>>
>

--00235447067494a47e047a61b7ad
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<br><div class=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=3D"b=
order-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; paddin=
g-left: 1ex;">Hello,<br><br>Below is the packet in UDP format, Could you pl=
ease help me on this to understand this scenario. Also attached is the comp=
lete traces.<br>
<br>INVITE <a href=3D"mailto:sip%3A9996214018210532@208.85.248.42" target=
=3D"_blank">sip:9996214018210532@208.85.248.42</a> SIP/2.0<br>
<br>Max-Forwards: 69<br><br>Session-Expires: 3600;refresher=3Duac<br><br>Su=
pported: timer, 100rel<br><br>To: &lt;<a href=3D"mailto:sip%3A9996214018210=
532@208.85.248.42" target=3D"_blank">sip:9996214018210532@208.85.248.42</a>=
&gt;<br>
<br>From: &lt;sip:8003223223@tmsash1-15-sip&gt;;tag=3D3468945569-250042<br>
<br>Call-ID: <a href=3D"mailto:7044988-3468945569-250036@mscasheq01.soundbi=
te.com" target=3D"_blank">7044988-3468945569-250036@mscasheq01.soundbite.co=
m</a><br><br>CSeq: 1 INVITE<br><br>Allow: INVITE, BYE, OPTIONS, CANCEL, ACK=
, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE<br>

<br>Via: SIP/2.0/UDP 8.15.244.10:5060;branch=3Dz9hG4bKf5de4d0e3c0860d122139=
e1e30a2995c<br><br>Contact: &lt;<a href=3D"http://sip:8003223223@8.15.244.1=
0:5060" target=3D"_blank">sip:8003223223@8.15.244.10:5060</a>&gt;<br><br>Ca=
ll-Info: &lt;sip:8.15.244.10&gt;;method=3D&quot;NOTIFY;Event=3Dtelephone-ev=
ent;Duration=3D1000&quot;<br>

<br>Content-Type: application/sdp<br><br>Content-Length: 124<br><br><br><br=
>v=3D0<br><br>o=3Dmscasheq01 547500705 547500705 IN IP4 8.15.244.10<br><br>=
s=3Dsip call<br><br>c=3DIN IP4 8.15.244.11<br><br>t=3D0 0<br><br>m=3Daudio =
19612 RTP/AVP 0<br>

<br>SIP/2.0 100 Trying<br><br>Via: SIP/2.0/UDP 8.15.244.10:5060;branch=3Dz9=
hG4bKf5de4d0e3c0860d122139e1e30a2995c<br><br>From: &lt;sip:8003223223@tmsas=
h1-15-sip&gt;;tag=3D3468945569-250042<br><br>To: &lt;<a href=3D"mailto:sip%=
3A9996214018210532@208.85.248.42" target=3D"_blank">sip:9996214018210532@20=
8.85.248.42</a>&gt;<br>

<br>Call-ID: <a href=3D"mailto:7044988-3468945569-250036@mscasheq01.soundbi=
te.com" target=3D"_blank">7044988-3468945569-250036@mscasheq01.soundbite.co=
m</a><br><br>CSeq: 1 INVITE<br><br>Server: Sansay VSX 2.1<br><br>Content-Le=
ngth: 0<br>
<br><br>
<br>SIP/2.0 180 Ringing<br><br>Via: SIP/2.0/UDP 8.15.244.10:5060;branch=3Dz=
9hG4bKf5de4d0e3c0860d122139e1e30a2995c<br><br>From: &lt;sip:8003223223@tmsa=
sh1-15-sip&gt;;tag=3D3468945569-250042<br><br>To: &lt;<a href=3D"mailto:sip=
%3A9996214018210532@208.85.248.42" target=3D"_blank">sip:9996214018210532@2=
08.85.248.42</a>&gt;;tag=3D228209687-tdb1145808424<br>

<br>Call-ID: <a href=3D"mailto:7044988-3468945569-250036@mscasheq01.soundbi=
te.com" target=3D"_blank">7044988-3468945569-250036@mscasheq01.soundbite.co=
m</a><br><br>CSeq: 1 INVITE<br><br>Server: Sansay VSX 2.1<br><br>Content-Le=
ngth: 0<br>
<br><br>
<br>SIP/2.0 200 OK<br><br>Via: SIP/2.0/UDP 8.15.244.10:5060;branch=3Dz9hG4b=
Kf5de4d0e3c0860d122139e1e30a2995c<br><br>From: &lt;sip:8003223223@tmsash1-1=
5-sip&gt;;tag=3D3468945569-250042<br><br>To: &lt;<a href=3D"mailto:sip%3A99=
96214018210532@208.85.248.42" target=3D"_blank">sip:9996214018210532@208.85=
.248.42</a>&gt;;tag=3D228209687-tdb1145808424<br>

<br>Call-ID: <a href=3D"mailto:7044988-3468945569-250036@mscasheq01.soundbi=
te.com" target=3D"_blank">7044988-3468945569-250036@mscasheq01.soundbite.co=
m</a><br><br>CSeq: 1 INVITE<br><br>Server: Sansay VSX 2.1<br><br>Contact: &=
lt;sip:9996214018210532@208.85.248.42:5060;transport=3Dudp&gt;<br>

<br>Content-Type: application/sdp<br><br>Content-Length:=A0=A0 158<br><br><=
br><br>v=3D0<br><br>o=3Dsansay-VSX 10 10 IN IP4 64.156.174.71<br><br>s=3Dse=
ssion controller<br><br>c=3DIN IP4 64.156.174.71<br><br>t=3D0 0<br><br>m=3D=
audio 10766 RTP/AVP 0<br>

<br>a=3Drtpmap:0 PCMU/8000<br><br>a=3Dptime:20<br><br>ACK sip:9996214018210=
532@208.85.248.42:5060;transport=3Dudp SIP/2.0<br><br>Max-Forwards: 69<br><=
br>To: &lt;<a href=3D"mailto:sip%3A9996214018210532@208.85.248.42" target=
=3D"_blank">sip:9996214018210532@208.85.248.42</a>&gt;;tag=3D228209687-tdb1=
145808424<br>

<br>From: &lt;sip:8003223223@tmsash1-15-sip&gt;;tag=3D3468945569-250042<br>=
<br>Call-ID: <a href=3D"mailto:7044988-3468945569-250036@mscasheq01.soundbi=
te.com" target=3D"_blank">7044988-3468945569-250036@mscasheq01.soundbite.co=
m</a><br>
<br>CSeq: 1 ACK<br>
<br>Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER=
, SUBSCRIBE, PRACK, UPDATE<br><br>Via: SIP/2.0/UDP 8.15.244.10:5060;branch=
=3Dz9hG4bK7208e1bfdd2962741dc3dee9d2576820<br><br>Contact: &lt;<a href=3D"h=
ttp://sip:8003223223@8.15.244.10:5060" target=3D"_blank">sip:8003223223@8.1=
5.244.10:5060</a>&gt;<br>

<br>Content-Length: 0<br><br><br>Thanks,<br>Nitin Kapoor<br><br><div class=
=3D"gmail_quote">2009/12/10 Christer Holmberg <span dir=3D"ltr">&lt;<a href=
=3D"mailto:christer.holmberg@ericsson.com" target=3D"_blank">christer.holmb=
erg@ericsson.com</a>&gt;</span><div>
<div></div><div class=3D"h5"><br>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><br>
Hi,<br>
<br>
It is possible.<br>
<br>
In theory this could be done with a single c- line, because SDP only talks =
about where media is to be sent - not where it comes from.<br>
<br>
In reality, however, I assume this is a case where each m- line has its own=
 c- line, each with a different IP address.<br>
<br>
Regards,<br>
<font color=3D"#888888"><br>
Christer<br>
</font><div><br>
<br>
-----Original Message-----<br>
From: <a href=3D"mailto:sip-bounces@ietf.org" target=3D"_blank">sip-bounces=
@ietf.org</a> on behalf of Nitin Kapoor<br>
Sent: Thu 12/10/2009 2:10 PM<br>
To: <a href=3D"mailto:sip@ietf.org" target=3D"_blank">sip@ietf.org</a><br>
Subject: [Sip] Two Media Stream in One call<br>
<br>
</div><div><div></div><div>Hello All,<br>
<br>
I am facing the issue with one of my customer where he is getting two media=
<br>
streams in one call and they are using SBC(Nextone)<br>
<br>
I checked the signaling traces and everything looks okay to me as well as<b=
r>
the RTP traces too.<br>
<br>
As per the traces one =A0RTP packet is going from 64.156.174.71 and the oth=
er<br>
is coming from 8.15.244.11.<br>
<br>
Could you please let me know is it possible where i can have the two media<=
br>
streams in one call when this is the not the conference call. And if this i=
s<br>
possible then what is the scenario where its possible.<br>
<br>
Please find the attached traces for the same.<br>
<br>
Thanks,<br>
Nitin Kapoor<br>
General Telecom<br>
<br>
</div></div></blockquote></div></div></div><br>
</blockquote></div><br>

--00235447067494a47e047a61b7ad--

From ron.even.tlv@gmail.com  Thu Dec 10 10:00:57 2009
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Christer Holmberg'" <christer.holmberg@ericsson.com>, "'Nitin Kapoor'" <nitinkapoorr@gmail.com>, <sip@ietf.org>
References: <c80c92d0912100510l3933eb4cud5c76c78413499c7@mail.gmail.com> <CA9998CD4A020D418654FCDEF4E707DF083CD445@esealmw113.eemea.ericsson.se>
In-Reply-To: <CA9998CD4A020D418654FCDEF4E707DF083CD445@esealmw113.eemea.ericsson.se>
Date: Thu, 10 Dec 2009 19:57:35 +0200
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Subject: Re: [Sip] Two Media Stream in One call
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Hi,
I looked at the SDP you send, you are seeing two RTP streams, one for each
direction. Since both sides talk than you see one RTP steam from 8.15.244.11
in one direction to port 10766 and one from 64.156.174.71 going to the other
direction to port 19612

Roni Even

> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
> Christer Holmberg
> Sent: Thursday, December 10, 2009 3:54 PM
> To: Nitin Kapoor; sip@ietf.org
> Subject: Re: [Sip] Two Media Stream in One call
> 
> 
> Hi,
> 
> It is possible.
> 
> In theory this could be done with a single c- line, because SDP only
> talks about where media is to be sent - not where it comes from.
> 
> In reality, however, I assume this is a case where each m- line has its
> own c- line, each with a different IP address.
> 
> Regards,
> 
> Christer
> 
> 
> -----Original Message-----
> From: sip-bounces@ietf.org on behalf of Nitin Kapoor
> Sent: Thu 12/10/2009 2:10 PM
> To: sip@ietf.org
> Subject: [Sip] Two Media Stream in One call
> 
> Hello All,
> 
> I am facing the issue with one of my customer where he is getting two
> media
> streams in one call and they are using SBC(Nextone)
> 
> I checked the signaling traces and everything looks okay to me as well
> as
> the RTP traces too.
> 
> As per the traces one  RTP packet is going from 64.156.174.71 and the
> other
> is coming from 8.15.244.11.
> 
> Could you please let me know is it possible where i can have the two
> media
> streams in one call when this is the not the conference call. And if
> this is
> possible then what is the scenario where its possible.
> 
> Please find the attached traces for the same.
> 
> Thanks,
> Nitin Kapoor
> General Telecom
> 
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


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Date: Thu, 10 Dec 2009 13:03:07 -0500
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From: Nitin Kapoor <nitinkapoorr@gmail.com>
To: Roni Even <ron.even.tlv@gmail.com>
Content-Type: multipart/alternative; boundary=0016368342589d700c047a639ece
Cc: sip@ietf.org, Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [Sip] Two Media Stream in One call
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--0016368342589d700c047a639ece
Content-Type: text/plain; charset=ISO-8859-1

Roni,

So is that the normal call.. I mean is it possible or you have seen any
issue into it..

Please help me on this.

Thanks,
Nitin Kapoor

2009/12/10 Roni Even <ron.even.tlv@gmail.com>

> Hi,
> I looked at the SDP you send, you are seeing two RTP streams, one for each
> direction. Since both sides talk than you see one RTP steam from
> 8.15.244.11
> in one direction to port 10766 and one from 64.156.174.71 going to the
> other
> direction to port 19612
>
> Roni Even
>
> > -----Original Message-----
> > From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
> > Christer Holmberg
> > Sent: Thursday, December 10, 2009 3:54 PM
> > To: Nitin Kapoor; sip@ietf.org
> > Subject: Re: [Sip] Two Media Stream in One call
> >
> >
> > Hi,
> >
> > It is possible.
> >
> > In theory this could be done with a single c- line, because SDP only
> > talks about where media is to be sent - not where it comes from.
> >
> > In reality, however, I assume this is a case where each m- line has its
> > own c- line, each with a different IP address.
> >
> > Regards,
> >
> > Christer
> >
> >
> > -----Original Message-----
> > From: sip-bounces@ietf.org on behalf of Nitin Kapoor
> > Sent: Thu 12/10/2009 2:10 PM
> > To: sip@ietf.org
> > Subject: [Sip] Two Media Stream in One call
> >
> > Hello All,
> >
> > I am facing the issue with one of my customer where he is getting two
> > media
> > streams in one call and they are using SBC(Nextone)
> >
> > I checked the signaling traces and everything looks okay to me as well
> > as
> > the RTP traces too.
> >
> > As per the traces one  RTP packet is going from 64.156.174.71 and the
> > other
> > is coming from 8.15.244.11.
> >
> > Could you please let me know is it possible where i can have the two
> > media
> > streams in one call when this is the not the conference call. And if
> > this is
> > possible then what is the scenario where its possible.
> >
> > Please find the attached traces for the same.
> >
> > Thanks,
> > Nitin Kapoor
> > General Telecom
> >
> > _______________________________________________
> > Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
>
>

--0016368342589d700c047a639ece
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

Roni,<br><br>So is that the normal call.. I mean is it possible or you have=
 seen any issue into it..<br><br>Please help me on this.<br><br>Thanks,<br>=
Nitin Kapoor<br><br><div class=3D"gmail_quote">2009/12/10 Roni Even <span d=
ir=3D"ltr">&lt;<a href=3D"mailto:ron.even.tlv@gmail.com">ron.even.tlv@gmail=
.com</a>&gt;</span><br>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Hi,<br>
I looked at the SDP you send, you are seeing two RTP streams, one for each<=
br>
direction. Since both sides talk than you see one RTP steam from 8.15.244.1=
1<br>
in one direction to port 10766 and one from 64.156.174.71 going to the othe=
r<br>
direction to port 19612<br>
<br>
Roni Even<br>
<div class=3D"im"><br>
&gt; -----Original Message-----<br>
&gt; From: <a href=3D"mailto:sip-bounces@ietf.org">sip-bounces@ietf.org</a>=
 [mailto:<a href=3D"mailto:sip-bounces@ietf.org">sip-bounces@ietf.org</a>] =
On Behalf Of<br>
</div><div><div></div><div class=3D"h5">&gt; Christer Holmberg<br>
&gt; Sent: Thursday, December 10, 2009 3:54 PM<br>
&gt; To: Nitin Kapoor; <a href=3D"mailto:sip@ietf.org">sip@ietf.org</a><br>
&gt; Subject: Re: [Sip] Two Media Stream in One call<br>
&gt;<br>
&gt;<br>
&gt; Hi,<br>
&gt;<br>
&gt; It is possible.<br>
&gt;<br>
&gt; In theory this could be done with a single c- line, because SDP only<b=
r>
&gt; talks about where media is to be sent - not where it comes from.<br>
&gt;<br>
&gt; In reality, however, I assume this is a case where each m- line has it=
s<br>
&gt; own c- line, each with a different IP address.<br>
&gt;<br>
&gt; Regards,<br>
&gt;<br>
&gt; Christer<br>
&gt;<br>
&gt;<br>
&gt; -----Original Message-----<br>
&gt; From: <a href=3D"mailto:sip-bounces@ietf.org">sip-bounces@ietf.org</a>=
 on behalf of Nitin Kapoor<br>
&gt; Sent: Thu 12/10/2009 2:10 PM<br>
&gt; To: <a href=3D"mailto:sip@ietf.org">sip@ietf.org</a><br>
&gt; Subject: [Sip] Two Media Stream in One call<br>
&gt;<br>
&gt; Hello All,<br>
&gt;<br>
&gt; I am facing the issue with one of my customer where he is getting two<=
br>
&gt; media<br>
&gt; streams in one call and they are using SBC(Nextone)<br>
&gt;<br>
&gt; I checked the signaling traces and everything looks okay to me as well=
<br>
&gt; as<br>
&gt; the RTP traces too.<br>
&gt;<br>
&gt; As per the traces one =A0RTP packet is going from 64.156.174.71 and th=
e<br>
&gt; other<br>
&gt; is coming from 8.15.244.11.<br>
&gt;<br>
&gt; Could you please let me know is it possible where i can have the two<b=
r>
&gt; media<br>
&gt; streams in one call when this is the not the conference call. And if<b=
r>
&gt; this is<br>
&gt; possible then what is the scenario where its possible.<br>
&gt;<br>
&gt; Please find the attached traces for the same.<br>
&gt;<br>
&gt; Thanks,<br>
&gt; Nitin Kapoor<br>
&gt; General Telecom<br>
&gt;<br>
</div></div>&gt; _______________________________________________<br>
&gt; Sip mailing list =A0<a href=3D"https://www.ietf.org/mailman/listinfo/s=
ip" target=3D"_blank">https://www.ietf.org/mailman/listinfo/sip</a><br>
&gt; This list is for NEW development of the core SIP Protocol<br>
&gt; Use <a href=3D"mailto:sip-implementors@cs.columbia.edu">sip-implemento=
rs@cs.columbia.edu</a> for questions on current sip<br>
&gt; Use <a href=3D"mailto:sipping@ietf.org">sipping@ietf.org</a> for new d=
evelopments on the application of sip<br>
<br>
</blockquote></div><br>

--0016368342589d700c047a639ece--

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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Nitin Kapoor'" <nitinkapoorr@gmail.com>
References: <c80c92d0912100510l3933eb4cud5c76c78413499c7@mail.gmail.com>	 <CA9998CD4A020D418654FCDEF4E707DF083CD445@esealmw113.eemea.ericsson.se>	 <4b213744.06e2660a.5909.37ce@mx.google.com> <c80c92d0912101003x69aaf584kcef098ba36ef4a46@mail.gmail.com>
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Date: Thu, 10 Dec 2009 20:33:10 +0200
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Cc: sip@ietf.org, 'Christer Holmberg' <christer.holmberg@ericsson.com>
Subject: Re: [Sip] Two Media Stream in One call
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This is a normal call, RTP is unidirectional and this is why you see two RTP
streams one from each side for each media channel.

Roni

 

From: Nitin Kapoor [mailto:nitinkapoorr@gmail.com] 
Sent: Thursday, December 10, 2009 8:03 PM
To: Roni Even
Cc: Christer Holmberg; sip@ietf.org
Subject: Re: [Sip] Two Media Stream in One call

 

Roni,

So is that the normal call.. I mean is it possible or you have seen any
issue into it..

Please help me on this.

Thanks,
Nitin Kapoor

2009/12/10 Roni Even <ron.even.tlv@gmail.com>

Hi,
I looked at the SDP you send, you are seeing two RTP streams, one for each
direction. Since both sides talk than you see one RTP steam from 8.15.244.11
in one direction to port 10766 and one from 64.156.174.71 going to the other
direction to port 19612

Roni Even


> -----Original Message-----
> From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of

> Christer Holmberg
> Sent: Thursday, December 10, 2009 3:54 PM
> To: Nitin Kapoor; sip@ietf.org
> Subject: Re: [Sip] Two Media Stream in One call
>
>
> Hi,
>
> It is possible.
>
> In theory this could be done with a single c- line, because SDP only
> talks about where media is to be sent - not where it comes from.
>
> In reality, however, I assume this is a case where each m- line has its
> own c- line, each with a different IP address.
>
> Regards,
>
> Christer
>
>
> -----Original Message-----
> From: sip-bounces@ietf.org on behalf of Nitin Kapoor
> Sent: Thu 12/10/2009 2:10 PM
> To: sip@ietf.org
> Subject: [Sip] Two Media Stream in One call
>
> Hello All,
>
> I am facing the issue with one of my customer where he is getting two
> media
> streams in one call and they are using SBC(Nextone)
>
> I checked the signaling traces and everything looks okay to me as well
> as
> the RTP traces too.
>
> As per the traces one  RTP packet is going from 64.156.174.71 and the
> other
> is coming from 8.15.244.11.
>
> Could you please let me know is it possible where i can have the two
> media
> streams in one call when this is the not the conference call. And if
> this is
> possible then what is the scenario where its possible.
>
> Please find the attached traces for the same.
>
> Thanks,
> Nitin Kapoor
> General Telecom
>

> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

 


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<div class=3DSection1>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>This is a normal call, RTP is unidirectional and this is =
why you
see two RTP streams one from each side for each media =
channel.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Roni<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<div style=3D'border:none;border-left:solid blue 1.5pt;padding:0in 0in =
0in 4.0pt'>

<div>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt =
0in 0in 0in'>

<p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> Nitin =
Kapoor
[mailto:nitinkapoorr@gmail.com] <br>
<b>Sent:</b> Thursday, December 10, 2009 8:03 PM<br>
<b>To:</b> Roni Even<br>
<b>Cc:</b> Christer Holmberg; sip@ietf.org<br>
<b>Subject:</b> Re: [Sip] Two Media Stream in One =
call<o:p></o:p></span></p>

</div>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'>Roni,<br>
<br>
So is that the normal call.. I mean is it possible or you have seen any =
issue
into it..<br>
<br>
Please help me on this.<br>
<br>
Thanks,<br>
Nitin Kapoor<o:p></o:p></p>

<div>

<p class=3DMsoNormal>2009/12/10 Roni Even &lt;<a
href=3D"mailto:ron.even.tlv@gmail.com">ron.even.tlv@gmail.com</a>&gt;<o:p=
></o:p></p>

<p class=3DMsoNormal>Hi,<br>
I looked at the SDP you send, you are seeing two RTP streams, one for =
each<br>
direction. Since both sides talk than you see one RTP steam from =
8.15.244.11<br>
in one direction to port 10766 and one from 64.156.174.71 going to the =
other<br>
direction to port 19612<br>
<br>
Roni Even<o:p></o:p></p>

<div>

<p class=3DMsoNormal><br>
&gt; -----Original Message-----<br>
&gt; From: <a =
href=3D"mailto:sip-bounces@ietf.org">sip-bounces@ietf.org</a>
[mailto:<a =
href=3D"mailto:sip-bounces@ietf.org">sip-bounces@ietf.org</a>] On
Behalf Of<o:p></o:p></p>

</div>

<div>

<div>

<p class=3DMsoNormal>&gt; Christer Holmberg<br>
&gt; Sent: Thursday, December 10, 2009 3:54 PM<br>
&gt; To: Nitin Kapoor; <a =
href=3D"mailto:sip@ietf.org">sip@ietf.org</a><br>
&gt; Subject: Re: [Sip] Two Media Stream in One call<br>
&gt;<br>
&gt;<br>
&gt; Hi,<br>
&gt;<br>
&gt; It is possible.<br>
&gt;<br>
&gt; In theory this could be done with a single c- line, because SDP =
only<br>
&gt; talks about where media is to be sent - not where it comes =
from.<br>
&gt;<br>
&gt; In reality, however, I assume this is a case where each m- line has =
its<br>
&gt; own c- line, each with a different IP address.<br>
&gt;<br>
&gt; Regards,<br>
&gt;<br>
&gt; Christer<br>
&gt;<br>
&gt;<br>
&gt; -----Original Message-----<br>
&gt; From: <a =
href=3D"mailto:sip-bounces@ietf.org">sip-bounces@ietf.org</a> on
behalf of Nitin Kapoor<br>
&gt; Sent: Thu 12/10/2009 2:10 PM<br>
&gt; To: <a href=3D"mailto:sip@ietf.org">sip@ietf.org</a><br>
&gt; Subject: [Sip] Two Media Stream in One call<br>
&gt;<br>
&gt; Hello All,<br>
&gt;<br>
&gt; I am facing the issue with one of my customer where he is getting =
two<br>
&gt; media<br>
&gt; streams in one call and they are using SBC(Nextone)<br>
&gt;<br>
&gt; I checked the signaling traces and everything looks okay to me as =
well<br>
&gt; as<br>
&gt; the RTP traces too.<br>
&gt;<br>
&gt; As per the traces one &nbsp;RTP packet is going from 64.156.174.71 =
and the<br>
&gt; other<br>
&gt; is coming from 8.15.244.11.<br>
&gt;<br>
&gt; Could you please let me know is it possible where i can have the =
two<br>
&gt; media<br>
&gt; streams in one call when this is the not the conference call. And =
if<br>
&gt; this is<br>
&gt; possible then what is the scenario where its possible.<br>
&gt;<br>
&gt; Please find the attached traces for the same.<br>
&gt;<br>
&gt; Thanks,<br>
&gt; Nitin Kapoor<br>
&gt; General Telecom<br>
&gt;<o:p></o:p></p>

</div>

</div>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'>&gt;
_______________________________________________<br>
&gt; Sip mailing list &nbsp;<a =
href=3D"https://www.ietf.org/mailman/listinfo/sip"
target=3D"_blank">https://www.ietf.org/mailman/listinfo/sip</a><br>
&gt; This list is for NEW development of the core SIP Protocol<br>
&gt; Use <a =
href=3D"mailto:sip-implementors@cs.columbia.edu">sip-implementors@cs.colu=
mbia.edu</a>
for questions on current sip<br>
&gt; Use <a href=3D"mailto:sipping@ietf.org">sipping@ietf.org</a> for =
new
developments on the application of sip<o:p></o:p></p>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

</div>

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From nitinkapoorr@gmail.com  Wed Dec 16 14:12:42 2009
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Date: Wed, 16 Dec 2009 17:12:23 -0500
Message-ID: <c80c92d0912161412v4cbabb8gbef30536e36a92c8@mail.gmail.com>
From: Nitin Kapoor <nitinkapoorr@gmail.com>
To: Roni Even <ron.even.tlv@gmail.com>
Content-Type: multipart/alternative; boundary=00163641704b0a63e7047adfcdcd
Cc: sip@ietf.org, Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [Sip] Two Media Stream in One call
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--00163641704b0a63e7047adfcdcd
Content-Type: text/plain; charset=ISO-8859-1

Hello Roni,

I still have the doubt.. the point you have mentioned that the the RTP is
the unidirectional and because of that it is showing me two RTP streams.

But if we are decoding the RTP packets and whenever i am trying to hear the
messages then i can see that both have the different conversion..which is
confusing me lot... because without CONFERENCE call how one call have two
different conversion or RTP stream  in one dialog...

Please correct me if i am wrong.

Thanks,
Nitin Kapoor

2009/12/10 Roni Even <ron.even.tlv@gmail.com>

>  This is a normal call, RTP is unidirectional and this is why you see two
> RTP streams one from each side for each media channel.
>
> Roni
>
>
>
> *From:* Nitin Kapoor [mailto:nitinkapoorr@gmail.com]
> *Sent:* Thursday, December 10, 2009 8:03 PM
> *To:* Roni Even
> *Cc:* Christer Holmberg; sip@ietf.org
>
> *Subject:* Re: [Sip] Two Media Stream in One call
>
>
>
> Roni,
>
> So is that the normal call.. I mean is it possible or you have seen any
> issue into it..
>
> Please help me on this.
>
> Thanks,
> Nitin Kapoor
>
> 2009/12/10 Roni Even <ron.even.tlv@gmail.com>
>
> Hi,
> I looked at the SDP you send, you are seeing two RTP streams, one for each
> direction. Since both sides talk than you see one RTP steam from
> 8.15.244.11
> in one direction to port 10766 and one from 64.156.174.71 going to the
> other
> direction to port 19612
>
> Roni Even
>
>
> > -----Original Message-----
> > From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
>
> > Christer Holmberg
> > Sent: Thursday, December 10, 2009 3:54 PM
> > To: Nitin Kapoor; sip@ietf.org
> > Subject: Re: [Sip] Two Media Stream in One call
> >
> >
> > Hi,
> >
> > It is possible.
> >
> > In theory this could be done with a single c- line, because SDP only
> > talks about where media is to be sent - not where it comes from.
> >
> > In reality, however, I assume this is a case where each m- line has its
> > own c- line, each with a different IP address.
> >
> > Regards,
> >
> > Christer
> >
> >
> > -----Original Message-----
> > From: sip-bounces@ietf.org on behalf of Nitin Kapoor
> > Sent: Thu 12/10/2009 2:10 PM
> > To: sip@ietf.org
> > Subject: [Sip] Two Media Stream in One call
> >
> > Hello All,
> >
> > I am facing the issue with one of my customer where he is getting two
> > media
> > streams in one call and they are using SBC(Nextone)
> >
> > I checked the signaling traces and everything looks okay to me as well
> > as
> > the RTP traces too.
> >
> > As per the traces one  RTP packet is going from 64.156.174.71 and the
> > other
> > is coming from 8.15.244.11.
> >
> > Could you please let me know is it possible where i can have the two
> > media
> > streams in one call when this is the not the conference call. And if
> > this is
> > possible then what is the scenario where its possible.
> >
> > Please find the attached traces for the same.
> >
> > Thanks,
> > Nitin Kapoor
> > General Telecom
> >
>
> > _______________________________________________
> > Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
>
>
>

--00163641704b0a63e7047adfcdcd
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

Hello Roni,<br><br>I still have the doubt.. the point you have mentioned th=
at the the RTP is the unidirectional and because of that it is showing me t=
wo RTP streams. <br><br>But if we are decoding the RTP packets and whenever=
 i am trying to hear the messages then i can see that both have the differe=
nt conversion..which is confusing me lot... because without CONFERENCE call=
 how one call have two different conversion or RTP stream=A0 in one dialog.=
..<br>
<br>Please correct me if i am wrong.<br><br>Thanks,<br>Nitin Kapoor<br><br>=
<div class=3D"gmail_quote">2009/12/10 Roni Even <span dir=3D"ltr">&lt;<a hr=
ef=3D"mailto:ron.even.tlv@gmail.com">ron.even.tlv@gmail.com</a>&gt;</span><=
br>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">








<div link=3D"blue" vlink=3D"purple" lang=3D"EN-US">

<div>

<p class=3D"MsoNormal"><span style=3D"font-size: 11pt; color: rgb(31, 73, 1=
25);">This is a normal call, RTP is unidirectional and this is why you
see two RTP streams one from each side for each media channel.</span></p>

<p class=3D"MsoNormal"><span style=3D"font-size: 11pt; color: rgb(31, 73, 1=
25);">Roni</span></p>

<p class=3D"MsoNormal"><span style=3D"font-size: 11pt; color: rgb(31, 73, 1=
25);">=A0</span></p>

<div style=3D"border-style: none none none solid; border-color: -moz-use-te=
xt-color -moz-use-text-color -moz-use-text-color blue; border-width: medium=
 medium medium 1.5pt; padding: 0in 0in 0in 4pt;">

<div>

<div style=3D"border-style: solid none none; border-color: rgb(181, 196, 22=
3) -moz-use-text-color -moz-use-text-color; border-width: 1pt medium medium=
; padding: 3pt 0in 0in;">

<p class=3D"MsoNormal"><b><span style=3D"font-size: 10pt;">From:</span></b>=
<span style=3D"font-size: 10pt;"> Nitin Kapoor
[mailto:<a href=3D"mailto:nitinkapoorr@gmail.com" target=3D"_blank">nitinka=
poorr@gmail.com</a>] <br>
<b>Sent:</b> Thursday, December 10, 2009 8:03 PM<br>
<b>To:</b> Roni Even<br>
<b>Cc:</b> Christer Holmberg; <a href=3D"mailto:sip@ietf.org" target=3D"_bl=
ank">sip@ietf.org</a><div><div></div><div class=3D"h5"><br>
<b>Subject:</b> Re: [Sip] Two Media Stream in One call</div></div></span></=
p>

</div>

</div><div><div></div><div class=3D"h5">

<p class=3D"MsoNormal">=A0</p>

<p class=3D"MsoNormal" style=3D"margin-bottom: 12pt;">Roni,<br>
<br>
So is that the normal call.. I mean is it possible or you have seen any iss=
ue
into it..<br>
<br>
Please help me on this.<br>
<br>
Thanks,<br>
Nitin Kapoor</p>

<div>

<p class=3D"MsoNormal">2009/12/10 Roni Even &lt;<a href=3D"mailto:ron.even.=
tlv@gmail.com" target=3D"_blank">ron.even.tlv@gmail.com</a>&gt;</p>

<p class=3D"MsoNormal">Hi,<br>
I looked at the SDP you send, you are seeing two RTP streams, one for each<=
br>
direction. Since both sides talk than you see one RTP steam from 8.15.244.1=
1<br>
in one direction to port 10766 and one from 64.156.174.71 going to the othe=
r<br>
direction to port 19612<br>
<br>
Roni Even</p>

<div>

<p class=3D"MsoNormal"><br>
&gt; -----Original Message-----<br>
&gt; From: <a href=3D"mailto:sip-bounces@ietf.org" target=3D"_blank">sip-bo=
unces@ietf.org</a>
[mailto:<a href=3D"mailto:sip-bounces@ietf.org" target=3D"_blank">sip-bounc=
es@ietf.org</a>] On
Behalf Of</p>

</div>

<div>

<div>

<p class=3D"MsoNormal">&gt; Christer Holmberg<br>
&gt; Sent: Thursday, December 10, 2009 3:54 PM<br>
&gt; To: Nitin Kapoor; <a href=3D"mailto:sip@ietf.org" target=3D"_blank">si=
p@ietf.org</a><br>
&gt; Subject: Re: [Sip] Two Media Stream in One call<br>
&gt;<br>
&gt;<br>
&gt; Hi,<br>
&gt;<br>
&gt; It is possible.<br>
&gt;<br>
&gt; In theory this could be done with a single c- line, because SDP only<b=
r>
&gt; talks about where media is to be sent - not where it comes from.<br>
&gt;<br>
&gt; In reality, however, I assume this is a case where each m- line has it=
s<br>
&gt; own c- line, each with a different IP address.<br>
&gt;<br>
&gt; Regards,<br>
&gt;<br>
&gt; Christer<br>
&gt;<br>
&gt;<br>
&gt; -----Original Message-----<br>
&gt; From: <a href=3D"mailto:sip-bounces@ietf.org" target=3D"_blank">sip-bo=
unces@ietf.org</a> on
behalf of Nitin Kapoor<br>
&gt; Sent: Thu 12/10/2009 2:10 PM<br>
&gt; To: <a href=3D"mailto:sip@ietf.org" target=3D"_blank">sip@ietf.org</a>=
<br>
&gt; Subject: [Sip] Two Media Stream in One call<br>
&gt;<br>
&gt; Hello All,<br>
&gt;<br>
&gt; I am facing the issue with one of my customer where he is getting two<=
br>
&gt; media<br>
&gt; streams in one call and they are using SBC(Nextone)<br>
&gt;<br>
&gt; I checked the signaling traces and everything looks okay to me as well=
<br>
&gt; as<br>
&gt; the RTP traces too.<br>
&gt;<br>
&gt; As per the traces one =A0RTP packet is going from 64.156.174.71 and th=
e<br>
&gt; other<br>
&gt; is coming from 8.15.244.11.<br>
&gt;<br>
&gt; Could you please let me know is it possible where i can have the two<b=
r>
&gt; media<br>
&gt; streams in one call when this is the not the conference call. And if<b=
r>
&gt; this is<br>
&gt; possible then what is the scenario where its possible.<br>
&gt;<br>
&gt; Please find the attached traces for the same.<br>
&gt;<br>
&gt; Thanks,<br>
&gt; Nitin Kapoor<br>
&gt; General Telecom<br>
&gt;</p>

</div>

</div>

<p class=3D"MsoNormal" style=3D"margin-bottom: 12pt;">&gt;
_______________________________________________<br>
&gt; Sip mailing list =A0<a href=3D"https://www.ietf.org/mailman/listinfo/s=
ip" target=3D"_blank">https://www.ietf.org/mailman/listinfo/sip</a><br>
&gt; This list is for NEW development of the core SIP Protocol<br>
&gt; Use <a href=3D"mailto:sip-implementors@cs.columbia.edu" target=3D"_bla=
nk">sip-implementors@cs.columbia.edu</a>
for questions on current sip<br>
&gt; Use <a href=3D"mailto:sipping@ietf.org" target=3D"_blank">sipping@ietf=
.org</a> for new
developments on the application of sip</p>

</div>

<p class=3D"MsoNormal">=A0</p>

</div></div></div>

</div>

</div>


</blockquote></div><br>

--00163641704b0a63e7047adfcdcd--

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Subject: [Sip] SIP over SCTP
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--0-1088832150-1261031703=:21045
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Hi All,

Can I ahve some frame of SIP which is coming over SCTP ? Or some pointer fr=
om where I can have it?/

Regards
Atul
=0A=0A=0A      The INTERNET now has a personality. YOURS! See your Yahoo! H=
omepage. http://in.yahoo.com/
--0-1088832150-1261031703=:21045
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<table cellspacing=3D"0" cellpadding=3D"0" border=3D"0" ><tr><td valign=3D"=
top" style=3D"font: inherit;">Hi All,<br><br>Can I ahve some frame of SIP w=
hich is coming over SCTP ? Or some pointer from where I can have it?/<br><b=
r>Regards<br>Atul<br></td></tr></table><br>=0A=0A=0A=0A      <!--1--><hr si=
ze=3D1></hr> =0AThe INTERNET now has a personality. YOURS! <a href=3D"http:=
//in.rd.yahoo.com/tagline_yyi_1/*http://in.yahoo.com/" target=3D"_blank">Se=
e your Yahoo! Homepage</a>.
--0-1088832150-1261031703=:21045--

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Hi, you have http://tools.ietf.org/html/rfc4168=0ABrds=0A=0AThomas=0A=0ALe =
17 d=E9c. 2009 =E0 07:35, atul garg <shivalk_80@yahoo.co.in> a =E9crit :=0A=
=0AHi All,=0A=0ACan I ahve some frame of SIP which is coming over SCTP ? Or=
 some pointer from where I can have it?/=0A=0ARegards=0AAtul=0A=0AThe INTER=
NET now has a personality. YOURS! See your Yahoo! Homepage.=0A_____________=
__________________________________=0ASip mailing list  https://www.ietf.org=
/mailman/listinfo/sip=0AThis list is for NEW development of the core SIP Pr=
otocol=0AUse sip-implementors@cs.columbia.edu for questions on current sip=
=0AUse sipping@ietf.org for new developments on the application of sip=0A=
=0A=0A=0A      
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<html><body bgcolor=3D"#FFFFFF"><div>Hi, you have&nbsp;<a href=3D"http://to=
ols.ietf.org/html/rfc4168"><a href=3D"http://tools.ietf.org/html/rfc4168">h=
ttp://tools.ietf.org/html/rfc4168</a></a></div><div>Brds<br><br>Thomas</div=
><div><br>Le 17 d=E9c. 2009 =E0 07:35, atul garg &lt;<a href=3D"mailto:shiv=
alk_80@yahoo.co.in">shivalk_80@yahoo.co.in</a>&gt; a =E9crit&nbsp;:<br><br>=
</div><div></div><blockquote type=3D"cite"><div><table cellspacing=3D"0" ce=
llpadding=3D"0" border=3D"0"><tbody><tr><td valign=3D"top" style=3D"font: i=
nherit;">Hi All,<br><br>Can I ahve some frame of SIP which is coming over S=
CTP ? Or some pointer from where I can have it?/<br><br>Regards<br>Atul<br>=
</td></tr></tbody></table><br>=0A=0A=0A=0A      <!--1--><hr size=3D"1"> =0A=
The INTERNET now has a personality. YOURS! <a href=3D"http://in.rd.yahoo.co=
m/tagline_yyi_1/*http://in.yahoo.com/" target=3D"_blank">See your Yahoo! Ho=
mepage</a>.</div></blockquote><blockquote type=3D"cite"><div><span>________=
_______________________________________</span><br><span>Sip mailing list &n=
bsp;<a href=3D"https://www.ietf.org/mailman/listinfo/sip"><a href=3D"https:=
//www.ietf.org/mailman/listinfo/sip">https://www.ietf.org/mailman/listinfo/=
sip</a></a></span><br><span>This list is for NEW development of the core SI=
P Protocol</span><br><span>Use <a href=3D"mailto:sip-implementors@cs.columb=
ia.edu">sip-implementors@cs.columbia.edu</a> for questions on current sip</=
span><br><span>Use <a href=3D"mailto:sipping@ietf.org">sipping@ietf.org</a>=
 for new developments on the application of sip</span></div></blockquote><b=
r>=0A=0A=0A=0A=0A      </body></html>
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Date: Thu, 17 Dec 2009 15:30:21 +0100
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From: Victor Pascual Avila <victor.pascual.avila@gmail.com>
To: atul garg <shivalk_80@yahoo.co.in>
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Subject: Re: [Sip] SIP over SCTP
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On Thu, Dec 17, 2009 at 7:35 AM, atul garg <shivalk_80@yahoo.co.in> wrote:
>
> Hi All,
>
> Can I ahve some frame of SIP which is coming over SCTP ? Or some pointer =
from where I can have it?/

Yes, basically be ready to receive over any stream, send over stream
zero and use unordered delivery.

http://tools.ietf.org/html/rfc4168

Cheers,
--
Victor Pascual =C3=81vila

From nitinkapoorr@gmail.com  Fri Dec 18 13:55:53 2009
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Date: Fri, 18 Dec 2009 16:55:32 -0500
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From: Nitin Kapoor <nitinkapoorr@gmail.com>
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Cc: Christer Holmberg <christer.holmberg@ericsson.com>
Subject: Re: [Sip] Two Media Stream in One call
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--0023545be1b4756261047b07cc81
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Hello,

Can anyone help me out please on this.

Thanks,
Nitin Kapoor

2009/12/16 Nitin Kapoor <nitinkapoorr@gmail.com>

> Hello Roni,
>
> I still have the doubt.. the point you have mentioned that the the RTP is
> the unidirectional and because of that it is showing me two RTP streams.
>
> But if we are decoding the RTP packets and whenever i am trying to hear the
> messages then i can see that both have the different conversion..which is
> confusing me lot... because without CONFERENCE call how one call have two
> different conversion or RTP stream  in one dialog...
>
> Please correct me if i am wrong.
>
>
> Thanks,
> Nitin Kapoor
>
> 2009/12/10 Roni Even <ron.even.tlv@gmail.com>
>
>>  This is a normal call, RTP is unidirectional and this is why you see two
>> RTP streams one from each side for each media channel.
>>
>> Roni
>>
>>
>>
>> *From:* Nitin Kapoor [mailto:nitinkapoorr@gmail.com]
>> *Sent:* Thursday, December 10, 2009 8:03 PM
>> *To:* Roni Even
>> *Cc:* Christer Holmberg; sip@ietf.org
>>
>> *Subject:* Re: [Sip] Two Media Stream in One call
>>
>>
>>
>> Roni,
>>
>> So is that the normal call.. I mean is it possible or you have seen any
>> issue into it..
>>
>> Please help me on this.
>>
>> Thanks,
>> Nitin Kapoor
>>
>> 2009/12/10 Roni Even <ron.even.tlv@gmail.com>
>>
>> Hi,
>> I looked at the SDP you send, you are seeing two RTP streams, one for each
>> direction. Since both sides talk than you see one RTP steam from
>> 8.15.244.11
>> in one direction to port 10766 and one from 64.156.174.71 going to the
>> other
>> direction to port 19612
>>
>> Roni Even
>>
>>
>> > -----Original Message-----
>> > From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
>>
>> > Christer Holmberg
>> > Sent: Thursday, December 10, 2009 3:54 PM
>> > To: Nitin Kapoor; sip@ietf.org
>> > Subject: Re: [Sip] Two Media Stream in One call
>> >
>> >
>> > Hi,
>> >
>> > It is possible.
>> >
>> > In theory this could be done with a single c- line, because SDP only
>> > talks about where media is to be sent - not where it comes from.
>> >
>> > In reality, however, I assume this is a case where each m- line has its
>> > own c- line, each with a different IP address.
>> >
>> > Regards,
>> >
>> > Christer
>> >
>> >
>> > -----Original Message-----
>> > From: sip-bounces@ietf.org on behalf of Nitin Kapoor
>> > Sent: Thu 12/10/2009 2:10 PM
>> > To: sip@ietf.org
>> > Subject: [Sip] Two Media Stream in One call
>> >
>> > Hello All,
>> >
>> > I am facing the issue with one of my customer where he is getting two
>> > media
>> > streams in one call and they are using SBC(Nextone)
>> >
>> > I checked the signaling traces and everything looks okay to me as well
>> > as
>> > the RTP traces too.
>> >
>> > As per the traces one  RTP packet is going from 64.156.174.71 and the
>> > other
>> > is coming from 8.15.244.11.
>> >
>> > Could you please let me know is it possible where i can have the two
>> > media
>> > streams in one call when this is the not the conference call. And if
>> > this is
>> > possible then what is the scenario where its possible.
>> >
>> > Please find the attached traces for the same.
>> >
>> > Thanks,
>> > Nitin Kapoor
>> > General Telecom
>> >
>>
>> > _______________________________________________
>> > Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
>> > This list is for NEW development of the core SIP Protocol
>> > Use sip-implementors@cs.columbia.edu for questions on current sip
>> > Use sipping@ietf.org for new developments on the application of sip
>>
>>
>>
>
>

--0023545be1b4756261047b07cc81
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

Hello,<br><br>Can anyone help me out please on this.<br><br>Thanks,<br>Niti=
n Kapoor<br><br><div class=3D"gmail_quote">2009/12/16 Nitin Kapoor <span di=
r=3D"ltr">&lt;<a href=3D"mailto:nitinkapoorr@gmail.com">nitinkapoorr@gmail.=
com</a>&gt;</span><br>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Hello Roni,<br><b=
r>I still have the doubt.. the point you have mentioned that the the RTP is=
 the unidirectional and because of that it is showing me two RTP streams. <=
br>
<br>But if we are decoding the RTP packets and whenever i am trying to hear=
 the messages then i can see that both have the different conversion..which=
 is confusing me lot... because without CONFERENCE call how one call have t=
wo different conversion or RTP stream=A0 in one dialog...<br>

<br>Please correct me if i am wrong.<div><div></div><div class=3D"h5"><br><=
br>Thanks,<br>Nitin Kapoor<br><br><div class=3D"gmail_quote">2009/12/10 Ron=
i Even <span dir=3D"ltr">&lt;<a href=3D"mailto:ron.even.tlv@gmail.com" targ=
et=3D"_blank">ron.even.tlv@gmail.com</a>&gt;</span><br>

<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">








<div link=3D"blue" vlink=3D"purple" lang=3D"EN-US">

<div>

<p class=3D"MsoNormal"><span style=3D"font-size: 11pt; color: rgb(31, 73, 1=
25);">This is a normal call, RTP is unidirectional and this is why you
see two RTP streams one from each side for each media channel.</span></p>

<p class=3D"MsoNormal"><span style=3D"font-size: 11pt; color: rgb(31, 73, 1=
25);">Roni</span></p>

<p class=3D"MsoNormal"><span style=3D"font-size: 11pt; color: rgb(31, 73, 1=
25);">=A0</span></p>

<div style=3D"border-style: none none none solid; border-color: -moz-use-te=
xt-color -moz-use-text-color -moz-use-text-color blue; border-width: medium=
 medium medium 1.5pt; padding: 0in 0in 0in 4pt;">

<div>

<div style=3D"border-style: solid none none; border-color: rgb(181, 196, 22=
3) -moz-use-text-color -moz-use-text-color; border-width: 1pt medium medium=
; padding: 3pt 0in 0in;">

<p class=3D"MsoNormal"><b><span style=3D"font-size: 10pt;">From:</span></b>=
<span style=3D"font-size: 10pt;"> Nitin Kapoor
[mailto:<a href=3D"mailto:nitinkapoorr@gmail.com" target=3D"_blank">nitinka=
poorr@gmail.com</a>] <br>
<b>Sent:</b> Thursday, December 10, 2009 8:03 PM<br>
<b>To:</b> Roni Even<br>
<b>Cc:</b> Christer Holmberg; <a href=3D"mailto:sip@ietf.org" target=3D"_bl=
ank">sip@ietf.org</a><div><div></div><div><br>
<b>Subject:</b> Re: [Sip] Two Media Stream in One call</div></div></span></=
p>

</div>

</div><div><div></div><div>

<p class=3D"MsoNormal">=A0</p>

<p class=3D"MsoNormal" style=3D"margin-bottom: 12pt;">Roni,<br>
<br>
So is that the normal call.. I mean is it possible or you have seen any iss=
ue
into it..<br>
<br>
Please help me on this.<br>
<br>
Thanks,<br>
Nitin Kapoor</p>

<div>

<p class=3D"MsoNormal">2009/12/10 Roni Even &lt;<a href=3D"mailto:ron.even.=
tlv@gmail.com" target=3D"_blank">ron.even.tlv@gmail.com</a>&gt;</p>

<p class=3D"MsoNormal">Hi,<br>
I looked at the SDP you send, you are seeing two RTP streams, one for each<=
br>
direction. Since both sides talk than you see one RTP steam from 8.15.244.1=
1<br>
in one direction to port 10766 and one from 64.156.174.71 going to the othe=
r<br>
direction to port 19612<br>
<br>
Roni Even</p>

<div>

<p class=3D"MsoNormal"><br>
&gt; -----Original Message-----<br>
&gt; From: <a href=3D"mailto:sip-bounces@ietf.org" target=3D"_blank">sip-bo=
unces@ietf.org</a>
[mailto:<a href=3D"mailto:sip-bounces@ietf.org" target=3D"_blank">sip-bounc=
es@ietf.org</a>] On
Behalf Of</p>

</div>

<div>

<div>

<p class=3D"MsoNormal">&gt; Christer Holmberg<br>
&gt; Sent: Thursday, December 10, 2009 3:54 PM<br>
&gt; To: Nitin Kapoor; <a href=3D"mailto:sip@ietf.org" target=3D"_blank">si=
p@ietf.org</a><br>
&gt; Subject: Re: [Sip] Two Media Stream in One call<br>
&gt;<br>
&gt;<br>
&gt; Hi,<br>
&gt;<br>
&gt; It is possible.<br>
&gt;<br>
&gt; In theory this could be done with a single c- line, because SDP only<b=
r>
&gt; talks about where media is to be sent - not where it comes from.<br>
&gt;<br>
&gt; In reality, however, I assume this is a case where each m- line has it=
s<br>
&gt; own c- line, each with a different IP address.<br>
&gt;<br>
&gt; Regards,<br>
&gt;<br>
&gt; Christer<br>
&gt;<br>
&gt;<br>
&gt; -----Original Message-----<br>
&gt; From: <a href=3D"mailto:sip-bounces@ietf.org" target=3D"_blank">sip-bo=
unces@ietf.org</a> on
behalf of Nitin Kapoor<br>
&gt; Sent: Thu 12/10/2009 2:10 PM<br>
&gt; To: <a href=3D"mailto:sip@ietf.org" target=3D"_blank">sip@ietf.org</a>=
<br>
&gt; Subject: [Sip] Two Media Stream in One call<br>
&gt;<br>
&gt; Hello All,<br>
&gt;<br>
&gt; I am facing the issue with one of my customer where he is getting two<=
br>
&gt; media<br>
&gt; streams in one call and they are using SBC(Nextone)<br>
&gt;<br>
&gt; I checked the signaling traces and everything looks okay to me as well=
<br>
&gt; as<br>
&gt; the RTP traces too.<br>
&gt;<br>
&gt; As per the traces one =A0RTP packet is going from 64.156.174.71 and th=
e<br>
&gt; other<br>
&gt; is coming from 8.15.244.11.<br>
&gt;<br>
&gt; Could you please let me know is it possible where i can have the two<b=
r>
&gt; media<br>
&gt; streams in one call when this is the not the conference call. And if<b=
r>
&gt; this is<br>
&gt; possible then what is the scenario where its possible.<br>
&gt;<br>
&gt; Please find the attached traces for the same.<br>
&gt;<br>
&gt; Thanks,<br>
&gt; Nitin Kapoor<br>
&gt; General Telecom<br>
&gt;</p>

</div>

</div>

<p class=3D"MsoNormal" style=3D"margin-bottom: 12pt;">&gt;
_______________________________________________<br>
&gt; Sip mailing list =A0<a href=3D"https://www.ietf.org/mailman/listinfo/s=
ip" target=3D"_blank">https://www.ietf.org/mailman/listinfo/sip</a><br>
&gt; This list is for NEW development of the core SIP Protocol<br>
&gt; Use <a href=3D"mailto:sip-implementors@cs.columbia.edu" target=3D"_bla=
nk">sip-implementors@cs.columbia.edu</a>
for questions on current sip<br>
&gt; Use <a href=3D"mailto:sipping@ietf.org" target=3D"_blank">sipping@ietf=
.org</a> for new
developments on the application of sip</p>

</div>

<p class=3D"MsoNormal">=A0</p>

</div></div></div>

</div>

</div>


</blockquote></div><br>
</div></div></blockquote></div><br>

--0023545be1b4756261047b07cc81--

From dworley@nortel.com  Fri Dec 18 14:01:35 2009
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On Fri, 2009-12-18 at 16:55 -0500, Nitin Kapoor wrote:
> Can anyone help me out please on this.

If we are to help, you will have to provide more information than you
have.  At a minimum, we need to see:

the SDP offer
the SDP answer
a description of each RTP stream you see, including its source address
and its destination address

Dale



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--000e0cd5c6ca45fee6047b07e7a8
Content-Type: text/plain; charset=ISO-8859-1

Dale,

I have the ethreal traces along with RTP stream... i can provide you
that..if that can help to debug the issue.

Thanks,
Nitin Kapoor

2009/12/18 Dale Worley <dworley@nortel.com>

> On Fri, 2009-12-18 at 16:55 -0500, Nitin Kapoor wrote:
> > Can anyone help me out please on this.
>
> If we are to help, you will have to provide more information than you
> have.  At a minimum, we need to see:
>
> the SDP offer
> the SDP answer
> a description of each RTP stream you see, including its source address
> and its destination address
>
> Dale
>
>
>

--000e0cd5c6ca45fee6047b07e7a8
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

Dale,<br><br>I have the ethreal traces along with RTP stream... i can provi=
de you that..if that can help to debug the issue.<br><br>Thanks,<br>Nitin K=
apoor<br><br><div class=3D"gmail_quote">2009/12/18 Dale Worley <span dir=3D=
"ltr">&lt;<a href=3D"mailto:dworley@nortel.com">dworley@nortel.com</a>&gt;<=
/span><br>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div class=3D"im"=
>On Fri, 2009-12-18 at 16:55 -0500, Nitin Kapoor wrote:<br>
&gt; Can anyone help me out please on this.<br>
<br>
</div>If we are to help, you will have to provide more information than you=
<br>
have. =A0At a minimum, we need to see:<br>
<br>
the SDP offer<br>
the SDP answer<br>
a description of each RTP stream you see, including its source address<br>
and its destination address<br>
<font color=3D"#888888"><br>
Dale<br>
<br>
<br>
</font></blockquote></div><br>

--000e0cd5c6ca45fee6047b07e7a8--

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I will be out of the office starting  12/19/2009 and will not return until
01/04/2010.



For Sun Microsystems related issues, please contact Arnaud Bellens
For 5560 IPT related issues, please contact Simon Binder

Otherwise, if the matter is of an urgent nature, please contact Stephen
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--00163641704b742a85047b3b0d8b
Content-Type: text/plain; charset=windows-1252
Content-Transfer-Encoding: quoted-printable

Hello Amit,

Okay... but whenever i am trying to decode the RTP packets and trying to
hear the conversation of this.. it seems to be a different conversation...
and which is confusing me a lot.

Also if you think after hearing the conversation of this call...the call is
still normal then could you please elaborate some evidence on which behalf
its look okay to you...

Thanks for your help in advance...

Thanks,
Nitin Kapoor

2009/12/21 Amit Vijayvargiya <amit.vijayvargiya@spicedigital.in>

>  Dear Nitin,
>
>
>
> In these traces every thing seems fine & it=92s a normal call. RTP Packet=
s
> are bidirectional
>
>
>
> 64.156.174.71(10766)  =DF--------=E0 8.15.244.11(19612)
>
>
>
> It should be a single conversation.
>
>
>
>
>
>
>
>  *Regards,*
> *Amit Vijayvargiya *
>   ------------------------------
>
> *From:* Nitin Kapoor [mailto:nitinkapoorr@gmail.com]
> *Sent:* Monday, December 21, 2009 10:16 AM
> *To:* Amit Vijayvargiya
>
> *Cc:* Christer Holmberg
> *Subject:* Re: [Sip] Two Media Stream in One call
>
>
>
> Hello Amit,
>
> Please find the attached traces...
>
> Thanks,
> Nitin Kapoor
>
> 2009/12/19 Amit Vijayvargiya <amit.vijayvargiya@spicedigital.in>
>
> Hi Nitin,
>
>
>
> Can you share the ethereal trace of RTP packets.
>
>
>
>
> *Thanks & Regards,*
>
> *Amit Vijayvargiya *
>   ------------------------------
>
> *From:* sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] *On Behalf Of =
*Nitin
> Kapoor
> *Sent:* Saturday, December 19, 2009 3:26 AM
>
>
> *To:* sip@ietf.org
> *Cc:* Christer Holmberg
>
> *Subject:* Re: [Sip] Two Media Stream in One call
>
>
>
> Hello,
>
> Can anyone help me out please on this.
>
> Thanks,
> Nitin Kapoor
>
> 2009/12/16 Nitin Kapoor <nitinkapoorr@gmail.com>
>
> Hello Roni,
>
> I still have the doubt.. the point you have mentioned that the the RTP is
> the unidirectional and because of that it is showing me two RTP streams.
>
> But if we are decoding the RTP packets and whenever i am trying to hear t=
he
> messages then i can see that both have the different conversion..which is
> confusing me lot... because without CONFERENCE call how one call have two
> different conversion or RTP stream  in one dialog...
>
> Please correct me if i am wrong.
>
>
>
> Thanks,
> Nitin Kapoor
>
> 2009/12/10 Roni Even <ron.even.tlv@gmail.com>
>
> This is a normal call, RTP is unidirectional and this is why you see two
> RTP streams one from each side for each media channel.
>
> Roni
>
>
>
> *From:* Nitin Kapoor [mailto:nitinkapoorr@gmail.com]
> *Sent:* Thursday, December 10, 2009 8:03 PM
> *To:* Roni Even
> *Cc:* Christer Holmberg; sip@ietf.org
>
>
> *Subject:* Re: [Sip] Two Media Stream in One call
>
>
>
> Roni,
>
> So is that the normal call.. I mean is it possible or you have seen any
> issue into it..
>
> Please help me on this.
>
> Thanks,
> Nitin Kapoor
>
> 2009/12/10 Roni Even <ron.even.tlv@gmail.com>
>
> Hi,
> I looked at the SDP you send, you are seeing two RTP streams, one for eac=
h
> direction. Since both sides talk than you see one RTP steam from
> 8.15.244.11
> in one direction to port 10766 and one from 64.156.174.71 going to the
> other
> direction to port 19612
>
> Roni Even
>
>
> > -----Original Message-----
> > From: sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] On Behalf Of
>
> > Christer Holmberg
> > Sent: Thursday, December 10, 2009 3:54 PM
> > To: Nitin Kapoor; sip@ietf.org
> > Subject: Re: [Sip] Two Media Stream in One call
> >
> >
> > Hi,
> >
> > It is possible.
> >
> > In theory this could be done with a single c- line, because SDP only
> > talks about where media is to be sent - not where it comes from.
> >
> > In reality, however, I assume this is a case where each m- line has its
> > own c- line, each with a different IP address.
> >
> > Regards,
> >
> > Christer
> >
> >
> > -----Original Message-----
> > From: sip-bounces@ietf.org on behalf of Nitin Kapoor
> > Sent: Thu 12/10/2009 2:10 PM
> > To: sip@ietf.org
> > Subject: [Sip] Two Media Stream in One call
> >
> > Hello All,
> >
> > I am facing the issue with one of my customer where he is getting two
> > media
> > streams in one call and they are using SBC(Nextone)
> >
> > I checked the signaling traces and everything looks okay to me as well
> > as
> > the RTP traces too.
> >
> > As per the traces one  RTP packet is going from 64.156.174.71 and the
> > other
> > is coming from 8.15.244.11.
> >
> > Could you please let me know is it possible where i can have the two
> > media
> > streams in one call when this is the not the conference call. And if
> > this is
> > possible then what is the scenario where its possible.
> >
> > Please find the attached traces for the same.
> >
> > Thanks,
> > Nitin Kapoor
> > General Telecom
> >
>
> > _______________________________________________
> > Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
>
>
>
>
>
>
>
>
>  ------------------------------
>
> This message and its attachments contain confidential information and may
> also contain legally privileged information. This message is intended sol=
ely
> for the named addressee. If you are not the addressee indicated in this
> message (or authorized to receive for addressee), you may not copy or
> deliver any part of this message or its attachments to anyone or use any
> part of this message or its attachments. Rather, you should permanently
> delete this message and its attachments (and all copies) from your system
> and kindly notify the sender by reply E-mail. Any content of this message
> and its attachments that does not relate to the official business of SPIC=
E
> DIGITAL or its affiliates and/or subsidiaries must be taken not to have b=
een
> sent or endorsed by any of them. Opinions and information in this E-mail
> that do not relate to the official business of SPICE DIGITAL., shall be
> understood as neither given nor endorsed by SPICE DIGITAL.. Any
> action/proceeding/opportunity upon the contents of this E-mail taken by n=
ot
> intended addressee shall attract liabilities both criminal and civil agai=
nst
> such addressee.
>
> E-mail communications are not private and no warranty is made that e-mail
> communications are timely, secure or free from computer virus or other
> defect.
>
>
>
> ------------------------------
> This message and its attachments contain confidential information and may
> also contain legally privileged information. This message is intended sol=
ely
> for the named addressee. If you are not the addressee indicated in this
> message (or authorized to receive for addressee), you may not copy or
> deliver any part of this message or its attachments to anyone or use any
> part of this message or its attachments. Rather, you should permanently
> delete this message and its attachments (and all copies) from your system
> and kindly notify the sender by reply E-mail. Any content of this message
> and its attachments that does not relate to the official business of SPIC=
E
> DIGITAL or its affiliates and/or subsidiaries must be taken not to have b=
een
> sent or endorsed by any of them. Opinions and information in this E-mail
> that do not relate to the official business of SPICE DIGITAL., shall be
> understood as neither given nor endorsed by SPICE DIGITAL.. Any
> action/proceeding/opportunity upon the contents of this E-mail taken by n=
ot
> intended addressee shall attract liabilities both criminal and civil agai=
nst
> such addressee.
>
> E-mail communications are not private and no warranty is made that e-mail
> communications are timely, secure or free from computer virus or other
> defect.
>

--00163641704b742a85047b3b0d8b
Content-Type: text/html; charset=windows-1252
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Hello Amit,<br><br>Okay... but whenever i am trying to decode the RTP packe=
ts and trying to hear the conversation of this.. it seems to be a different=
 conversation... and which is confusing me a lot.<br><br>Also if you think =
after hearing the conversation of this call...the call is still normal then=
 could you please elaborate some evidence on which behalf its look okay to =
you...<br>
<br>Thanks for your help in advance...<br><br>Thanks,<br>Nitin Kapoor<br><b=
r><div class=3D"gmail_quote">2009/12/21 Amit Vijayvargiya <span dir=3D"ltr"=
>&lt;<a href=3D"mailto:amit.vijayvargiya@spicedigital.in">amit.vijayvargiya=
@spicedigital.in</a>&gt;</span><br>
<blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid rgb(204, =
204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">





<div link=3D"blue" vlink=3D"blue" lang=3D"EN-US">
<div>
<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial" size=3D"2"><span=
 style=3D"font-size: 10pt; font-family: Arial; color: navy;">Dear Nitin,</s=
pan></font></p>
<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial" size=3D"2"><span=
 style=3D"font-size: 10pt; font-family: Arial; color: navy;">=A0</span></fo=
nt></p>
<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial" size=3D"2"><span=
 style=3D"font-size: 10pt; font-family: Arial; color: navy;">In these trace=
s every thing seems fine &amp; it=92s a normal call. RTP Packets are bidire=
ctional
</span></font></p>
<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial" size=3D"2"><span=
 style=3D"font-size: 10pt; font-family: Arial; color: navy;">=A0</span></fo=
nt></p>
<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial" size=3D"2"><span=
 style=3D"font-size: 10pt; font-family: Arial; color: navy;">64.156.174.71(=
10766)=A0
</span></font><font color=3D"navy" face=3D"Wingdings" size=3D"2"><span styl=
e=3D"font-size: 10pt; font-family: Wingdings; color: navy;">=DF</span></fon=
t><font color=3D"navy" face=3D"Arial" size=3D"2"><span style=3D"font-size: =
10pt; font-family: Arial; color: navy;">--------</span></font><font color=
=3D"navy" face=3D"Wingdings" size=3D"2"><span style=3D"font-size: 10pt; fon=
t-family: Wingdings; color: navy;">=E0</span></font><font color=3D"navy" fa=
ce=3D"Arial" size=3D"2"><span style=3D"font-size: 10pt; font-family: Arial;=
 color: navy;">
 8.15.244.11(19612) =A0=A0</span></font></p>
<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial" size=3D"2"><span=
 style=3D"font-size: 10pt; font-family: Arial; color: navy;">=A0</span></fo=
nt></p>
<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial" size=3D"2"><span=
 style=3D"font-size: 10pt; font-family: Arial; color: navy;">It should be a=
 single conversation.
</span></font></p>
<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial" size=3D"2"><span=
 style=3D"font-size: 10pt; font-family: Arial; color: navy;">=A0</span></fo=
nt></p>
<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial" size=3D"2"><span=
 style=3D"font-size: 10pt; font-family: Arial; color: navy;">=A0</span></fo=
nt></p>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom: 12pt;"><font color=3D"black"=
 face=3D"Verdana" size=3D"2"><span style=3D"font-size: 10pt; font-family: V=
erdana; color: black;"><br>
<br>
</span></font></p>
<p class=3D"MsoNormal" style=3D"margin-bottom: 12pt;"><b><font color=3D"bla=
ck" face=3D"Verdana" size=3D"2"><span style=3D"font-size: 10pt; font-family=
: Verdana; color: black; font-weight: bold;">Regards,</span></font></b><fon=
t color=3D"black" face=3D"Verdana" size=3D"2"><span style=3D"font-size: 10p=
t; font-family: Verdana; color: black;"><br>

<b><span style=3D"font-weight: bold;">Amit Vijayvargiya </span></b></span><=
/font></p>
</div>
<div>
<div class=3D"MsoNormal" style=3D"text-align: center;" align=3D"center"><fo=
nt face=3D"Times New Roman" size=3D"3"><span style=3D"font-size: 12pt;">
<hr width=3D"100%" align=3D"center" size=3D"2">
</span></font></div>
<p class=3D"MsoNormal"><b><font face=3D"Tahoma" size=3D"2"><span style=3D"f=
ont-size: 10pt; font-family: Tahoma; font-weight: bold;">From:</span></font=
></b><font face=3D"Tahoma" size=3D"2"><span style=3D"font-size: 10pt; font-=
family: Tahoma;"> Nitin Kapoor [mailto:<a href=3D"mailto:nitinkapoorr@gmail=
.com" target=3D"_blank">nitinkapoorr@gmail.com</a>]
<br>
<b><span style=3D"font-weight: bold;">Sent:</span></b> Monday, December 21,=
 2009 10:16 AM<br>
<b><span style=3D"font-weight: bold;">To:</span></b> Amit Vijayvargiya<div>=
<div></div><div class=3D"h5"><br>
<b><span style=3D"font-weight: bold;">Cc:</span></b> Christer Holmberg<br>
<b><span style=3D"font-weight: bold;">Subject:</span></b> Re: [Sip] Two Med=
ia Stream in One call</div></div></span></font></p>
</div><div><div></div><div class=3D"h5">
<p class=3D"MsoNormal"><font face=3D"Times New Roman" size=3D"3"><span styl=
e=3D"font-size: 12pt;">=A0</span></font></p>
<p class=3D"MsoNormal" style=3D"margin-bottom: 12pt;"><font face=3D"Times N=
ew Roman" size=3D"3"><span style=3D"font-size: 12pt;">Hello Amit,<br>
<br>
Please find the attached traces...<br>
<br>
Thanks,<br>
Nitin Kapoor</span></font></p>
<div>
<p class=3D"MsoNormal"><font face=3D"Times New Roman" size=3D"3"><span styl=
e=3D"font-size: 12pt;">2009/12/19 Amit Vijayvargiya &lt;<a href=3D"mailto:a=
mit.vijayvargiya@spicedigital.in" target=3D"_blank">amit.vijayvargiya@spice=
digital.in</a>&gt;</span></font></p>

<div link=3D"blue" vlink=3D"blue">
<div>
<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial" size=3D"2"><span=
 style=3D"font-size: 10pt; font-family: Arial; color: navy;">Hi Nitin,</spa=
n></font></p>
<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial" size=3D"2"><span=
 style=3D"font-size: 10pt; font-family: Arial; color: navy;">=A0</span></fo=
nt></p>
<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial" size=3D"2"><span=
 style=3D"font-size: 10pt; font-family: Arial; color: navy;">Can you share =
the ethereal trace of RTP packets.
</span></font></p>
<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial" size=3D"2"><span=
 style=3D"font-size: 10pt; font-family: Arial; color: navy;">=A0</span></fo=
nt></p>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom: 12pt;"><font color=3D"black"=
 face=3D"Verdana" size=3D"2"><span style=3D"font-size: 10pt; font-family: V=
erdana; color: black;"><br>
<b><span style=3D"font-weight: bold;">Thanks &amp; Regards,</span></b><br>
<br>
<b><span style=3D"font-weight: bold;">Amit Vijayvargiya </span></b></span><=
/font></p>
</div>
<div>
<div class=3D"MsoNormal" style=3D"text-align: center;" align=3D"center"><fo=
nt face=3D"Times New Roman" size=3D"3"><span style=3D"font-size: 12pt;">
<hr width=3D"100%" align=3D"center" size=3D"2">
</span></font></div>
<p class=3D"MsoNormal"><b><font face=3D"Tahoma" size=3D"2"><span style=3D"f=
ont-size: 10pt; font-family: Tahoma; font-weight: bold;">From:</span></font=
></b><font face=3D"Tahoma" size=3D"2"><span style=3D"font-size: 10pt; font-=
family: Tahoma;">
<a href=3D"mailto:sip-bounces@ietf.org" target=3D"_blank">sip-bounces@ietf.=
org</a> [mailto:<a href=3D"mailto:sip-bounces@ietf.org" target=3D"_blank">s=
ip-bounces@ietf.org</a>]
<b><span style=3D"font-weight: bold;">On Behalf Of </span></b>Nitin Kapoor<=
br>
<b><span style=3D"font-weight: bold;">Sent:</span></b> Saturday, December 1=
9, 2009 3:26 AM</span></font></p>
<div>
<p class=3D"MsoNormal"><font face=3D"Tahoma" size=3D"2"><span style=3D"font=
-size: 10pt; font-family: Tahoma;"><br>
<b><span style=3D"font-weight: bold;">To:</span></b> <a href=3D"mailto:sip@=
ietf.org" target=3D"_blank">
sip@ietf.org</a><br>
<b><span style=3D"font-weight: bold;">Cc:</span></b> Christer Holmberg</spa=
n></font></p>
</div>
<div>
<div>
<p class=3D"MsoNormal"><b><font face=3D"Tahoma" size=3D"2"><span style=3D"f=
ont-size: 10pt; font-family: Tahoma; font-weight: bold;">Subject:</span></f=
ont></b><font face=3D"Tahoma" size=3D"2"><span style=3D"font-size: 10pt; fo=
nt-family: Tahoma;"> Re: [Sip] Two Media Stream in
 One call</span></font></p>
</div>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal"><font face=3D"Times New Roman" size=3D"3"><span styl=
e=3D"font-size: 12pt;">=A0</span></font></p>
<p class=3D"MsoNormal" style=3D"margin-bottom: 12pt;"><font face=3D"Times N=
ew Roman" size=3D"3"><span style=3D"font-size: 12pt;">Hello,<br>
<br>
Can anyone help me out please on this.<br>
<br>
Thanks,<br>
Nitin Kapoor</span></font></p>
<div>
<p class=3D"MsoNormal"><font face=3D"Times New Roman" size=3D"3"><span styl=
e=3D"font-size: 12pt;">2009/12/16 Nitin Kapoor &lt;<a href=3D"mailto:nitink=
apoorr@gmail.com" target=3D"_blank">nitinkapoorr@gmail.com</a>&gt;</span></=
font></p>

<p class=3D"MsoNormal"><font face=3D"Times New Roman" size=3D"3"><span styl=
e=3D"font-size: 12pt;">Hello Roni,<br>
<br>
I still have the doubt.. the point you have mentioned that the the RTP is t=
he unidirectional and because of that it is showing me two RTP streams.
<br>
<br>
But if we are decoding the RTP packets and whenever i am trying to hear the=
 messages then i can see that both have the different conversion..which is =
confusing me lot... because without CONFERENCE call how one call have two d=
ifferent conversion or RTP stream=A0
 in one dialog...<br>
<br>
Please correct me if i am wrong.</span></font></p>
<div>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom: 12pt;"><font face=3D"Times N=
ew Roman" size=3D"3"><span style=3D"font-size: 12pt;"><br>
<br>
Thanks,<br>
Nitin Kapoor</span></font></p>
<div>
<p class=3D"MsoNormal"><font face=3D"Times New Roman" size=3D"3"><span styl=
e=3D"font-size: 12pt;">2009/12/10 Roni Even &lt;<a href=3D"mailto:ron.even.=
tlv@gmail.com" target=3D"_blank">ron.even.tlv@gmail.com</a>&gt;</span></fon=
t></p>
<div link=3D"blue" vlink=3D"purple">
<div>
<p class=3D"MsoNormal"><font color=3D"#1f497d" face=3D"Times New Roman" siz=
e=3D"2"><span style=3D"font-size: 11pt; color: rgb(31, 73, 125);">This is a=
 normal call, RTP is unidirectional and this is why you see two RTP
 streams one from each side for each media channel.</span></font></p>
<p class=3D"MsoNormal"><font color=3D"#1f497d" face=3D"Times New Roman" siz=
e=3D"2"><span style=3D"font-size: 11pt; color: rgb(31, 73, 125);">Roni</spa=
n></font></p>
<p class=3D"MsoNormal"><font color=3D"#1f497d" face=3D"Times New Roman" siz=
e=3D"2"><span style=3D"font-size: 11pt; color: rgb(31, 73, 125);">=A0</span=
></font></p>
<div style=3D"border-style: none none none solid; border-color: -moz-use-te=
xt-color -moz-use-text-color -moz-use-text-color blue; border-width: medium=
 medium medium 1.5pt; padding: 0in 0in 0in 4pt;">
<div>
<div style=3D"border-style: solid none none; border-color: -moz-use-text-co=
lor; border-width: 1pt medium medium; padding: 3pt 0in 0in;">
<p class=3D"MsoNormal"><b><font face=3D"Times New Roman" size=3D"2"><span s=
tyle=3D"font-size: 10pt; font-weight: bold;">From:</span></font></b><font s=
ize=3D"2"><span style=3D"font-size: 10pt;"> Nitin Kapoor [mailto:<a href=3D=
"mailto:nitinkapoorr@gmail.com" target=3D"_blank">nitinkapoorr@gmail.com</a=
>]
<br>
<b><span style=3D"font-weight: bold;">Sent:</span></b> Thursday, December 1=
0, 2009 8:03 PM<br>
<b><span style=3D"font-weight: bold;">To:</span></b> Roni Even<br>
<b><span style=3D"font-weight: bold;">Cc:</span></b> Christer Holmberg; <a =
href=3D"mailto:sip@ietf.org" target=3D"_blank">
sip@ietf.org</a></span></font></p>
<div>
<div>
<p class=3D"MsoNormal"><font face=3D"Times New Roman" size=3D"2"><span styl=
e=3D"font-size: 10pt;"><br>
<b><span style=3D"font-weight: bold;">Subject:</span></b> Re: [Sip] Two Med=
ia Stream in One call</span></font></p>
</div>
</div>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal"><font face=3D"Times New Roman" size=3D"3"><span styl=
e=3D"font-size: 12pt;">=A0</span></font></p>
<p class=3D"MsoNormal" style=3D"margin-bottom: 12pt;"><font face=3D"Times N=
ew Roman" size=3D"3"><span style=3D"font-size: 12pt;">Roni,<br>
<br>
So is that the normal call.. I mean is it possible or you have seen any iss=
ue into it..<br>
<br>
Please help me on this.<br>
<br>
Thanks,<br>
Nitin Kapoor</span></font></p>
<div>
<p class=3D"MsoNormal"><font face=3D"Times New Roman" size=3D"3"><span styl=
e=3D"font-size: 12pt;">2009/12/10 Roni Even &lt;<a href=3D"mailto:ron.even.=
tlv@gmail.com" target=3D"_blank">ron.even.tlv@gmail.com</a>&gt;</span></fon=
t></p>
<p class=3D"MsoNormal"><font face=3D"Times New Roman" size=3D"3"><span styl=
e=3D"font-size: 12pt;">Hi,<br>
I looked at the SDP you send, you are seeing two RTP streams, one for each<=
br>
direction. Since both sides talk than you see one RTP steam from 8.15.244.1=
1<br>
in one direction to port 10766 and one from 64.156.174.71 going to the othe=
r<br>
direction to port 19612<br>
<br>
Roni Even</span></font></p>
<div>
<p class=3D"MsoNormal"><font face=3D"Times New Roman" size=3D"3"><span styl=
e=3D"font-size: 12pt;"><br>
&gt; -----Original Message-----<br>
&gt; From: <a href=3D"mailto:sip-bounces@ietf.org" target=3D"_blank">sip-bo=
unces@ietf.org</a> [mailto:<a href=3D"mailto:sip-bounces@ietf.org" target=
=3D"_blank">sip-bounces@ietf.org</a>] On Behalf Of</span></font></p>
</div>
<div>
<div>
<p class=3D"MsoNormal"><font face=3D"Times New Roman" size=3D"3"><span styl=
e=3D"font-size: 12pt;">&gt; Christer Holmberg<br>
&gt; Sent: Thursday, December 10, 2009 3:54 PM<br>
&gt; To: Nitin Kapoor; <a href=3D"mailto:sip@ietf.org" target=3D"_blank">si=
p@ietf.org</a><br>
&gt; Subject: Re: [Sip] Two Media Stream in One call<br>
&gt;<br>
&gt;<br>
&gt; Hi,<br>
&gt;<br>
&gt; It is possible.<br>
&gt;<br>
&gt; In theory this could be done with a single c- line, because SDP only<b=
r>
&gt; talks about where media is to be sent - not where it comes from.<br>
&gt;<br>
&gt; In reality, however, I assume this is a case where each m- line has it=
s<br>
&gt; own c- line, each with a different IP address.<br>
&gt;<br>
&gt; Regards,<br>
&gt;<br>
&gt; Christer<br>
&gt;<br>
&gt;<br>
&gt; -----Original Message-----<br>
&gt; From: <a href=3D"mailto:sip-bounces@ietf.org" target=3D"_blank">sip-bo=
unces@ietf.org</a> on behalf of Nitin Kapoor<br>
&gt; Sent: Thu 12/10/2009 2:10 PM<br>
&gt; To: <a href=3D"mailto:sip@ietf.org" target=3D"_blank">sip@ietf.org</a>=
<br>
&gt; Subject: [Sip] Two Media Stream in One call<br>
&gt;<br>
&gt; Hello All,<br>
&gt;<br>
&gt; I am facing the issue with one of my customer where he is getting two<=
br>
&gt; media<br>
&gt; streams in one call and they are using SBC(Nextone)<br>
&gt;<br>
&gt; I checked the signaling traces and everything looks okay to me as well=
<br>
&gt; as<br>
&gt; the RTP traces too.<br>
&gt;<br>
&gt; As per the traces one =A0RTP packet is going from 64.156.174.71 and th=
e<br>
&gt; other<br>
&gt; is coming from 8.15.244.11.<br>
&gt;<br>
&gt; Could you please let me know is it possible where i can have the two<b=
r>
&gt; media<br>
&gt; streams in one call when this is the not the conference call. And if<b=
r>
&gt; this is<br>
&gt; possible then what is the scenario where its possible.<br>
&gt;<br>
&gt; Please find the attached traces for the same.<br>
&gt;<br>
&gt; Thanks,<br>
&gt; Nitin Kapoor<br>
&gt; General Telecom<br>
&gt;</span></font></p>
</div>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom: 12pt;"><font face=3D"Times N=
ew Roman" size=3D"3"><span style=3D"font-size: 12pt;">&gt; ________________=
_______________________________<br>
&gt; Sip mailing list =A0<a href=3D"https://www.ietf.org/mailman/listinfo/s=
ip" target=3D"_blank">https://www.ietf.org/mailman/listinfo/sip</a><br>
&gt; This list is for NEW development of the core SIP Protocol<br>
&gt; Use <a href=3D"mailto:sip-implementors@cs.columbia.edu" target=3D"_bla=
nk">sip-implementors@cs.columbia.edu</a> for questions on current sip<br>
&gt; Use <a href=3D"mailto:sipping@ietf.org" target=3D"_blank">sipping@ietf=
.org</a> for new developments on the application of sip</span></font></p>
</div>
<p class=3D"MsoNormal"><font face=3D"Times New Roman" size=3D"3"><span styl=
e=3D"font-size: 12pt;">=A0</span></font></p>
</div>
</div>
</div>
</div>
</div>
</div>
<p class=3D"MsoNormal"><font face=3D"Times New Roman" size=3D"3"><span styl=
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</div>
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</div>
<p class=3D"MsoNormal"><font face=3D"Times New Roman" size=3D"3"><span styl=
e=3D"font-size: 12pt;">=A0</span></font></p>
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</div>
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<hr width=3D"100%" align=3D"center" size=3D"2">
</span></font></div>
<p class=3D"MsoNormal"><font color=3D"gray" face=3D"Arial" size=3D"1"><span=
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From gupta.vineet@gmail.com  Tue Dec 22 03:57:59 2009
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From: Vineet Gupta <gupta.vineet@gmail.com>
To: sip@ietf.org
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Subject: [Sip] Significance of port in maddr parameter.
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Hi,

Can someone let me know why exactly following statement is present in RFC
3261:

*"The maddr parameter indicates the server address to be contacted for this
user, overriding any address derived from the host field.  When an maddr
parameter is present, the port and transport components of the URI apply to
the address indicated in the maddr parameter value"*

To be specific:
a) What exactly can go wrong on using the port in maddr parameter?
b) In what are situations where port present in Req-URI should be used
rather than port present in maddr parameter?
c) And in what situation port in maddr parameter should be used?

Regards,
Vineet.

--0016e64d95feb14f8a047b4fe998
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Hi,<div><br></div><div>Can someone let me know why exactly following statem=
ent is present in RFC 3261:</div><div><br></div><div><div><i>&quot;The madd=
r parameter indicates the server address to be=A0contacted for this user, o=
verriding any address derived from=A0the host field. =A0<b>When an maddr pa=
rameter is present, the port=A0and transport components of the URI apply to=
 the address=A0indicated in the maddr parameter value</b>&quot;</i></div>
</div><div><br></div><div>To be specific:</div><div>a) What exactly can go =
wrong on using the port in maddr parameter?</div><div>b) In what are situat=
ions where port present in Req-URI should be used rather than port present =
in maddr parameter?=A0</div>
<div>c) And in what situation port in maddr parameter should be used?</div>=
<div>=A0</div><div>Regards,</div><div>Vineet.</div>

--0016e64d95feb14f8a047b4fe998--

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Hi Vineet,
As per RFC 3261, I don't see a placeholder for port number in the
maddr-param.
The ABNF syntax for maddr goes as follows:

maddr-param       =3D  "maddr=3D" host
host             =3D  hostname / IPv4address / IPv6reference

please explain on what do you mean by 'port in maddr param'



Thanks,
Best Regards,
Rekha

=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
=3D=3D=3D=3D=3D=3D=3D=3D=3D


Today's Topics:

   1. Significance of port in maddr parameter. (Vineet Gupta)


----------------------------------------------------------------------

Message: 1
Date: Tue, 22 Dec 2009 17:27:40 +0530
From: Vineet Gupta <gupta.vineet@gmail.com>
Subject: [Sip] Significance of port in maddr parameter.
To: sip@ietf.org
Message-ID:
	<98abe9690912220357t62881642l2230f190cfb64888@mail.gmail.com>
Content-Type: text/plain; charset=3D"iso-8859-1"

Hi,

Can someone let me know why exactly following statement is present in
RFC
3261:

*"The maddr parameter indicates the server address to be contacted for
this
user, overriding any address derived from the host field.  When an maddr
parameter is present, the port and transport components of the URI apply
to
the address indicated in the maddr parameter value"*

To be specific:
a) What exactly can go wrong on using the port in maddr parameter?
b) In what are situations where port present in Req-URI should be used
rather than port present in maddr parameter?
c) And in what situation port in maddr parameter should be used?

Regards,
Vineet.
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<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT FACE=3D"Consolas">Hi =
Vineet,</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT FACE=3D"Consolas">As per RFC =
3261, I</FONT></SPAN><SPAN LANG=3D"en-us"> <FONT =
FACE=3D"Consolas">don&#8217;t</FONT></SPAN><SPAN LANG=3D"en-us"><FONT =
FACE=3D"Consolas"> see a placeholder for</FONT></SPAN><SPAN =
LANG=3D"en-us"> <FONT COLOR=3D"#FF0000" FACE=3D"Consolas">port =
number</FONT></SPAN><SPAN LANG=3D"en-us"><FONT FACE=3D"Consolas"> in the =
maddr-param.</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT FACE=3D"Consolas">The =
ABNF</FONT></SPAN><SPAN LANG=3D"en-us"> <FONT =
FACE=3D"Consolas">syntax</FONT></SPAN><SPAN LANG=3D"en-us"><FONT =
FACE=3D"Consolas"> for maddr goes as follows:</FONT></SPAN><SPAN =
LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT COLOR=3D"#FF0000" =
FACE=3D"Consolas">maddr-param&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
=3D&nbsp; &quot;maddr=3D&quot; host</FONT></SPAN><SPAN =
LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT COLOR=3D"#FF0000" =
FACE=3D"Consolas">host&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp; =3D&nbsp; hostname / IPv4address / =
IPv6reference</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT FACE=3D"Consolas">please explain =
on what do you mean by 'port in maddr param'</FONT></SPAN><SPAN =
LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"></SPAN><SPAN LANG=3D"en-us"><FONT =
FACE=3D"Calibri">Thanks,</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT FACE=3D"Calibri">Best =
Regards,</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT =
FACE=3D"Calibri">Rekha</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT =
FACE=3D"Calibri">=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN><SPAN =
LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT FACE=3D"Consolas">Today's =
Topics:</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT FACE=3D"Consolas">&nbsp;&nbsp; =
1. Significance of port in maddr parameter. (Vineet =
Gupta)</FONT></SPAN></P>
<BR>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT =
FACE=3D"Consolas">-------------------------------------------------------=
---------------</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT FACE=3D"Consolas">Message: =
1</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT FACE=3D"Consolas">Date: Tue, 22 =
Dec 2009 17:27:40 +0530</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT FACE=3D"Consolas">From: Vineet =
Gupta &lt;gupta.vineet@gmail.com&gt;</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT FACE=3D"Consolas">Subject: [Sip] =
Significance of port in maddr parameter.</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT FACE=3D"Consolas">To: =
sip@ietf.org</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT =
FACE=3D"Consolas">Message-ID:</FONT></SPAN></P>

<P DIR=3DLTR><SPAN =
LANG=3D"en-us">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
FACE=3D"Consolas">&lt;98abe9690912220357t62881642l2230f190cfb64888@mail.g=
mail.com&gt;</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT FACE=3D"Consolas">Content-Type: =
text/plain; charset=3D&quot;iso-8859-1&quot;</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT =
FACE=3D"Consolas">Hi,</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT FACE=3D"Consolas">Can someone =
let me know why exactly following statement is present in =
RFC</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT =
FACE=3D"Consolas">3261:</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT FACE=3D"Consolas">*&quot;The =
maddr parameter indicates the server address to be contacted for =
this</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT FACE=3D"Consolas">user, =
overriding any address derived from the host field.&nbsp; When an =
maddr</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT FACE=3D"Consolas">parameter is =
present, the port and transport components of the URI apply =
to</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT FACE=3D"Consolas">the address =
indicated in the maddr parameter value&quot;*</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT FACE=3D"Consolas">To be =
specific:</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT FACE=3D"Consolas">a) What =
exactly can go wrong on using the port in maddr =
parameter?</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT FACE=3D"Consolas">b) In what are =
situations where port present in Req-URI should be =
used</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT FACE=3D"Consolas">rather than =
port present in maddr parameter?</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT FACE=3D"Consolas">c) And in what =
situation port in maddr parameter should be used?</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT =
FACE=3D"Consolas">Regards,</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT =
FACE=3D"Consolas">Vineet.</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT FACE=3D"Consolas">-------------- =
next part --------------</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT FACE=3D"Consolas">An HTML =
attachment was scrubbed...</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT FACE=3D"Consolas">URL: &lt;<A =
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<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT =
FACE=3D"Consolas">------------------------------</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT =
FACE=3D"Consolas">_______________________________________________</FONT><=
/SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT FACE=3D"Consolas">Sip mailing =
list&nbsp; <A =
HREF=3D"https://www.ietf.org/mailman/listinfo/sip">https://www.ietf.org/m=
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<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT FACE=3D"Consolas">This list is =
for NEW development of the core SIP Protocol</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT FACE=3D"Consolas">Use =
sip-implementors@cs.columbia.edu for questions on current =
sip</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT FACE=3D"Consolas">Use =
sipping@ietf.org for new developments on the application of =
sip</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT FACE=3D"Consolas">End of Sip =
Digest, Vol 64, Issue 10</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT =
FACE=3D"Consolas">***********************************</FONT></SPAN></P>

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From abhatnagar192006@gmail.com  Tue Dec 22 23:55:23 2009
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Date: Wed, 23 Dec 2009 08:25:02 +0030
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From: aayush bhatnagar <abhatnagar192006@gmail.com>
To: "Sreerekha Shenoy (sresheno)" <sresheno@cisco.com>
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Cc: sip@ietf.org
Subject: Re: [Sip] Significance of port in maddr parameter.
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Vineet,

The maddr parameter can be present in the Via header and in the request URI.

If present in the Via header, it signifies that the client is sending
the request to a multicast receiver. Refer to section 18.1.1 of 3261
for this.

It may be present in the Request URI...which means that the entry of
the request uri is added as the only entry in the target set for the
request.

As for the port, it is not present in the maddr paramater. It is
present in the sent-by parameter, or defaults to 5060 in case the
sent-by parameter is absent.

Hope this helps. For more info of maddr parameter usage, read sections
16.4, 16.5, 18.1.1, 18.1.2 and 19.1.1 of RFC 3261.

On 23/12/2009, Sreerekha Shenoy (sresheno) <sresheno@cisco.com> wrote:
> Hi Vineet,
> As per RFC 3261, I don't see a placeholder for port number in the
> maddr-param.
> The ABNF syntax for maddr goes as follows:
>
> maddr-param       =  "maddr=" host
> host             =  hostname / IPv4address / IPv6reference
>
> please explain on what do you mean by 'port in maddr param'
>
>
>
> Thanks,
> Best Regards,
> Rekha
>
> ========================================================================
> =========
>
>
> Today's Topics:
>
>    1. Significance of port in maddr parameter. (Vineet Gupta)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Tue, 22 Dec 2009 17:27:40 +0530
> From: Vineet Gupta <gupta.vineet@gmail.com>
> Subject: [Sip] Significance of port in maddr parameter.
> To: sip@ietf.org
> Message-ID:
> 	<98abe9690912220357t62881642l2230f190cfb64888@mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi,
>
> Can someone let me know why exactly following statement is present in
> RFC
> 3261:
>
> *"The maddr parameter indicates the server address to be contacted for
> this
> user, overriding any address derived from the host field.  When an maddr
> parameter is present, the port and transport components of the URI apply
> to
> the address indicated in the maddr parameter value"*
>
> To be specific:
> a) What exactly can go wrong on using the port in maddr parameter?
> b) In what are situations where port present in Req-URI should be used
> rather than port present in maddr parameter?
> c) And in what situation port in maddr parameter should be used?
>
> Regards,
> Vineet.
> -------------- next part --------------
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>
> ------------------------------
>
> _______________________________________________
> Sip mailing list  https://www.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>
> End of Sip Digest, Vol 64, Issue 10
> ***********************************
>


-- 
aayush
http://in.linkedin.com/in/abhatnagar19
======
