[SIPForum-discussion] difference in the sdp for the calls !!

Tomasz Karbowski tom170 at o2.pl
Mon Dec 5 07:03:48 UTC 2011


Dear Virajith,

1 & 2. In m line 18 = G729, 0=G711U, 8=G711A, 4=?, 106 & 100 probably DTMF  
as you can see it in a=rtpmap and a=fmtp.
3. Its telling you that audio can be sent and received (when you do HOLD  
it can be sendonly and on the other side recvonly, in video scenarios it  
can be for ex. inactive where ports are negotiated for video but the  
stream should be not sent)

BR,
Tom

Dnia 03-12-2011 o 07:50:30 virajith  <vir356 at rediffmail.com> napisał(a):

>
> Dear all,
>
> I have&nbsp; a query in relation to&nbsp; 2 different types of&nbsp; sdp  
> received in&nbsp; the INVITE from&nbsp; a sip provider for&nbsp;  
> incoming calls.
>
> Scenario 1
> =========&gt;
>
> =======sdp&nbsp; for few&nbsp; calls==========
>
> v=0
> o=- 2341457 1 IN IP4 60.234.18.158
> s=-
> c=IN IP4 60.234.18.158
> t=0 0
> m=audio 30006 RTP/AVP 8 0 18 100
> a=fmtp:18 annexb=no
> a=fmtp:100 0-15
> a=ptime:20
> a=rtpmap:100 telephone-event/8000
>
> ===============================
>
>
> Scenario 2
> =========&gt;
>
> ===========sdp for other calls===========
>
> v=0
> o=default 1322426685 1322426685 IN IP4 60.234.18.158
> s=-
> c=IN IP4 60.234.18.158
> t=0 0
> m=audio 30002 RTP/AVP 18 106 4 8 0
> a=sendrecv
> a=rtpmap:106 telephone-event/8000
> a=fmtp:106 0-15
> a=maxptime:90
>
> ============================================
>
>
> Now the questions I have is the following...
>
> 1&gt;what are the codecs&nbsp; and the dtmf&nbsp; methods being  
> advertised in scenario 1 ?
>
> 2&gt;what are the codecs&nbsp; and the dtmf&nbsp; methods being  
> advertised in scenario 2 ?
>
> 3&gt;what is a=sendrecv ?
>
> 4&gt;also anything else that you can point out&nbsp; that is different  
> in the&nbsp;&nbsp;&nbsp; 2 sdps&nbsp;&nbsp; ?
>
>
> Thanks,
> Vir
>
>
>
>


-- 
Pozdrawiam
Tomasz Karbowski




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