Hi Anthony, Excuse me if I'm wrong but inbound-late-negotiation must be used proxy_media as I see in documentation. I don't want to proxy media because of some issues with MOH or 3-way conferencing. Also I want to exclude media codecs that are supported only in pass-trough mode. Let mi give you an example: SDP from caller v=0 o=- 1 2 IN IP4 192.168.20.193 s=CounterPath eyeBeam 1.5 c=IN IP4 192.168.40.81 t=0 0 m=audio 56888 RTP/AVP 100 106 97 105 98 3 101 a=fmtp:101 0-15 a=rtpmap:100 SPEEX/16000 a=rtpmap:106 SPEEX-FEC/16000 a=rtpmap:97 SPEEX/8000 a=rtpmap:105 SPEEX-FEC/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv a=x-rtp-session-id:61940567309B49E8909127E1393A966E m=video 46378 RTP/AVP 125 115 34 a=fmtp:125 profile-level-id=42e015; max-br=4000; max-mbps=19800 a=fmtp:115 QCIF=1 MAXBR=4520 a=fmtp:34 QCIF=1 MAXBR=4520 a=rtpmap:125 H264/90000 a=rtpmap:115 H263-1998/90000 a=rtpmap:34 H263/90000 a=sendrecv a=x-rtp-session-id:E8244E608F65445BA183BBE641C5DF3C a=nortpproxy:yes SDP from Freeswitch to called v=0 o=FreeSWITCH 6228154995644782318 4633980766357417433 IN IP4 10.10.10.10 s=FreeSWITCH c=IN IP4 10.10.10.10 t=0 0 m=audio 26920 RTP/AVP 3 101 13 * a=rtpmap:3 GSM/8000* a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 So we offer *only first* in codec preference list from called wich is normal receives SIP/2.0 488 Not Acceptable Here *called suports - PCMA,PCMU,iLBC * Codec preference to this vars.xml we have witch is used in provile: <X-PRE-PROCESS cmd="set" data="all_codec_prefs=GSM,speex@32000h @20i,speex@16000h@20i,speex@8000h@20i,*iLBC@30i,PCMA,PCMU*,G729"/> also we have in profile: <param name="inbound-codec-negotiation" value="greedy"/> <param name="disable-transcoding" value="true"/> In dialplan I've set: <action application="set" data="inbound-late-negotiation"/> <action application="set" data="inherit_codec=true"/> About my second question: Why I should parse variable_switch_r_sdp: [v=0 o=- 6 2 IN IP4 192.168.20.193 s=CounterPath eyeBeam 1.5 c=IN IP4 192.168.40.81 t=0 0 m=audio 60642 RTP/AVP 100 106 97 105 98 3 101 a=rtpmap:100 SPEEX/16000 a=rtpmap:106 SPEEX-FEC/16000 a=rtpmap:97 SPEEX/8000 a=rtpmap:105 SPEEX-FEC/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=x-rtp-session-id:0674790128EB43ACB8C7F55829BCFF14 m=video 44938 RTP/AVP 125 115 34 a=rtpmap:125 H264/90000 a=fmtp:125 profile-level-id=42e015; max-br=4000; max-mbps=19800 a=rtpmap:115 H263-1998/90000 a=fmtp:115 QCIF=1 MAXBR=4520 a=rtpmap:34 H263/90000 a=fmtp:34 QCIF=1 MAXBR=4520 a=x-rtp-session-id:D1EAE543055746AC9C03006B91ADE6DF a=nortpproxy:yes ] In FS core this parse is already done I'm sure in much more intelligent way. It can be exported as a variable like a absolute codec string I think. Thanks again. 2009/2/12 Anthony Minessale <anthony.minessale@gmail.com>
the entire sdp is available as a variable (route the call to the info app to see the variables) so if you have inbound-late-negotiation set to true on the sip profile then you can use a regex or a script to set absolute_codec string before you answer.
On Thu, Feb 12, 2009 at 8:06 AM, ivdreg ivdreg <ivdreg@gmail.com> wrote:
Hi all,
Can I ask 2 questions about codec negotiation:
1. Is it possible Freeswitch to work negotiate codecs between two phones as it is described below. INVITE from A with some codecs in SDP ---> Freeswitch rewrites codec preference according absolute_codec_string but exclude all codecs not offered by A ----> INVITE to B with rewrited SDP.
example: from A SDP:PCMA,PCMU,SPEEX ----> absolute_codec_string=G722,PCMU,PCMA,GSM ----> to B SDP: PCMU,PCMA
2. Can I get codec list in INVITE with mod_perl for example or via xml_curl without processing SDP variable (switch_r_sdp). It will be useful to be in format that absolute_codec_string variable takes.
Thanks
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