[Freeswitch-users] Codec negotiation questions
ivdreg ivdreg
ivdreg at gmail.com
Fri Feb 13 08:50:00 PST 2009
Hi Anthony,
I'm not sure that you understood the problem. As it shown bellow the offered
codec in leg B contains only one codec (first matched in codec preference
list for this profile). Is there way to offer in leg B not only first codec
but all codecs that exists in INVITE in leg A that matches codec preference
list. If not is the only way is to parse SDP and set absolute_codec_string
manualy?
Regards
2009/2/13 Anthony Minessale <anthony.minessale at gmail.com>
> yes you are wrong.
>
> inbound-late-negotiation setting delays the codec negotiation until the
> instant audio is needed.
> It is not tied to inbound-proxy-media.
>
>
> This allows the call to come into the dialplan before any codec negotiation
> is done giving you a chance to look at the SDP before the negotiation takes
> place and insert an absolute_codec string essentially letting you chose
> unique codec preferences per inbound call.
>
> >>> Why I should parse variable_switch_r_sdp
>
> Well....you must parse it because it's you who cares about what it says, as
> described above it lets you peek at the sdp and enforce a unique set of
> codec prefs per call.
>
>
>
>
> On Fri, Feb 13, 2009 at 8:57 AM, ivdreg ivdreg <ivdreg at gmail.com> wrote:
>
>> 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 at 32000h
>> @20i,speex at 16000h@20i,speex at 8000h@20i,*iLBC at 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 at 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 at 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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>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
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>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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>>>
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>>>
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>>
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>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
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> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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>
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