[Freeswitch-users] rtpmap question PCMU/PCMA

[Mark Campbell-Smith]

Ahha... I found why it wasn't working.

I have media_bypass active and it appears this does not work when
media_bypass is active.

Is there any reason for this?

Why can't FreeSWITCH modify the signalling and then let the users
negotiate which codec to use?  If I modify the codec list to be one
that is not supported by the server, then its my fault.





On Wed, Jun 2, 2010 at 5:03 PM, Mark Campbell-Smith
<mcampbellsmith at gmail.com> wrote:
> Thanks Brian.  I didnt think it was mandatory.  As my client is not
> accepting this sdp format, can I change it?
>
> I've tried with
> <action application="export" data="codec_string=PCMA,PCMU"/>
> and
> <action application="export" data="absolute_codec_string=PCMA,PCMU"/>
> and
>            <action application="export"
> data="sip_append_audio_sdp=a=rtpmap:8 PCMA/8000"/>
>            <action application="export"
> data="sip_append_audio_sdp=a=rtpmap:0 PCMU/8000"/>
>
> and have late negotiation active, but nothing is changed in the SDP.
>
> Basically I want change
>   v=0
>   o=Sippy 142967340 3687727620696226321 IN IP4 80.232.37.178
>   s=-
>   t=0 0
>   m=audio 36824 RTP/AVP 8 0 18 101
>   c=IN IP4 213.50.91.3
>   a=fmtp:18 annexb=yes
>   a=rtpmap:101 telephone-event/8000
>   a=fmtp:101 0-15
>   a=sqn: 0
>   a=cdsc: 1 audio RTP/AVP  8
>   a=cdsc: 2 image udptl t38
>   a=cpar: a=T38FaxUdpEC:t38UDPRedundancy
>   a=cpar: a=T38FaxVersion:0
>   a=cpar: a=T38MaxBitRate:14400
>
> to something like:
>   v=0
>   o=Sippy 142967340 3687727620696226321 IN IP4 80.232.37.178
>   s=-
>   t=0 0
>   m=audio 36824 RTP/AVP 8 0 18 101
>   c=IN IP4 213.50.91.3
>   a=rtpmap:101 telephone-event/8000
>   a=fmtp:101 0-15
>   a=rtpmap:8 PCMA/8000
>   a=rtpmap:0 PCMU/8000
>
> Is there an 'easy' way to do this?
> Thanks
>
>
> On Wed, Jun 2, 2010 at 4:21 PM, Brian West <brian at freeswitch.org> wrote:
>> No it is NOT mandatory to include them.  Only when they are on defined codec numbers.
>>
>> /b
>>
>> On Jun 2, 2010, at 9:14 AM, Mark Campbell-Smith wrote:
>>
>>> Coming back to my original question:
>>> Is it mandatory to always include the rtpmap details for PCMU/PCMA
>>> codes?  For example something like 'a=rtpmap:8 PCMA/8000' and
>>> 'a=rtpmap:0 PCMU/8000'?
>>>
>>> Is it possible to add the end of the SDP parameters by doing something like:
>>>         <extension name="fix_phonzo" continue="true">
>>>                  <condition field="caller_id_number" expression="123456">
>>>                      <action application="export"
>>> data="sip_append_audio_sdp=a=rtpmap:8 PCMA/8000"/>
>>>                     <action application="export"
>>> data="sip_append_audio_sdp=a=rtpmap:0 PCMU/8000"/>
>>>                  </condition>
>>>            </extension>
>>>
>>> Can this be done anywhere in the dialplan as long as its before the
>>> bridge command?
>>
>>
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>