[Freeswitch-users] Codecs in SDP

[Michael Collins]

Well, PCMA and PCMU are included. Note this line:

m=audio 17262 RTP/AVP 8 0 101 13

8 = PCMA, 0 = PCMU

Not sure why there's no GSM in there but if you review the entire debug log
for the call in question it may yield a clue.

FYI, IIRC this variable will give you what you are looking for on the extra
SDP lines:

https://wiki.freeswitch.org/wiki/Channel_Variables#verbose_sdp

-MC


On Thu, Jun 5, 2014 at 10:48 AM, Ričardas Stoma <ricardas.stoma at gmail.com>
wrote:

> I have created the following extension:
>
> <extension name="my_test">
>   <condition>
>     <action application="bridge"
> data="{absolute_codec_string='PCMA,PCMU,GSM'}sofia/external/
> 1001 at 192.168.1.78"/>
>     <action application="hangup"/>
>   </condition>
> </extension>
>
> then i call to this extension, freeswitch dials to 1001 at 192.168.1.78 and
> when i check the content of SDP, included in a SIP packet,  i see something
> like this:
>
> v=0
> o=FreeSWITCH 1401972565 1401972566 IN IP4 192.168.1.65
> s=FreeSWITCH
> c=IN IP4 192.168.1.65
> t=0 0
> m=audio 17262 RTP/AVP 8 0 101 13
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
>
> why my codecs from 'absolute_codec_string' are not included in SDP? I
> would expect to see these lines included:
>
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:3 GSM/8000
>
>
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