[Freeswitch-users] Codec negotiation questions
Anthony Minessale
anthony.minessale at gmail.com
Fri Feb 13 09:21:25 PST 2009
As i have already answered, no, it does not do what you want automaticly,
the only way to influence codec negotiation is the way i have described.
parsing the sdp string allows you to set absolute_codec_string going both
ways.
if you set it before you answer the channel with late negotiation enabled it
will influence the codecs accepted on the inbound call.
it you set it on the b leg either by using export instead of set on the a
leg or putting it in {} in the dial string it controlls what codecs are
offered in the outbound invite.
On Fri, Feb 13, 2009 at 10:50 AM, ivdreg ivdreg <ivdreg at gmail.com> wrote:
> 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
>>>>>
>>>>> _______________________________________________
>>>>> Freeswitch-users mailing list
>>>>> Freeswitch-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> 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
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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
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>>
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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
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
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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