[Freeswitch-users] Disable annexb for g729

[Mike Rice]

I got it from the following URL.

https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation

---------- Forwarded message ----------
From: Michael Jerris <mike at jerris.com>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Cc:
Date: Tue, 17 May 2016 16:45:16 -0400
Subject: Re: [Freeswitch-users] Disable annexb for g729
It was spread about pretty widely when we went to 1.4 but thats a couple
years ago now.  It should be documented on the wiki but may have fallen off
during the transition to confluence as 1.2 is quite old.  Where did you see
docs for the sip_ var?

On Tue, May 17, 2016 at 3:51 PM, Mike Rice <mrice0118 at gmail.com> wrote:

> Thanks Michael! That did the trick. Did I miss that somewhere?
>
> From: Michael Jerris <mike at jerris.com>
>
> <action application="export" data="rtp_append_audio_sdp=a=fmtp:18
> annexb=no"/>
>
> between 1.2 and 1.4 the var got renamed from sip_ to rtp_
>
>
> On Tue, May 17, 2016 at 10:57 AM, Mike Rice <mrice0118 at gmail.com> wrote:
>
>> We have a carrier the mandates that annex b be disabled on the invites to
>> them. I have added the following to the default dialplan but nothing seems
>> to change the invite on the B leg to the carrier.
>>
>> <extension name="disable-annexB" continue="true">
>>   <condition field="${switch_r_sdp}" expression="/(.*)(m=audio \d+
>> RTP\/AVP)(.*)( 18 )(.*)/s">
>>      <action application="export" data="sip_append_audio_sdp=a=fmtp:18
>> annexb=no"/>
>>   </condition>
>> </extension>
>>
>> <action application="export" data="sip_append_audio_sdp=a=fmtp:18
>> annexb=no"/>
>>
>> and
>>
>> <action application="bridge" data="{sip_append_audio_sdp=a=fmtp:18
>>
>> annexb=no,absolute_codec_string=^^:G729:PCMU:PCMA}sofia/gateway/carrierGW/$1"/>
>>
>>
>> inbound-late-negotiation is set to true. The logs show that it is
>> exporting:
>>
>> [DEBUG] switch_channel.c:1267 EXPORT (export_vars)
>> [sip_h_Diversion]=[<sip:XXXXXXXXXX at 10.10.X.X>;reason=unavailable]
>> EXECUTE sofia/internal/1000 at 10.10.X.X
>> bridge(sofia/gateway/lo7f/XXXXXXXXXX)
>> [DEBUG] switch_channel.c:1221 sofia/internal/1000 at 10.10.X.X
>> EXPORTING[export_vars] [sip_append_audio_sdp]=[a=fmtp:18 annexb=no] to event
>> [DEBUG] switch_channel.c:1221 sofia/internal/1000 at 10.10.X.X
>> EXPORTING[export_vars] [sip_h_Diversion]=[<sip:XXXXXXXXXX at 10.10.X.X>;reason=unavailable]
>> to event
>>
>> The SDP does not reflect:
>>
>> Local SDP:
>> v=0
>> o=FreeSWITCH IN IP4 10.10.X.X
>> s=FreeSWITCH
>> c=IN IP4 10.10.X.X
>> t=0 0
>> m=audio 20022 RTP/AVP 18 0 8 101 13
>> a=rtpmap:18 G729/8000
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=ptime:20
>> a=sendrecv
>>
>> Any help would be greatly appreciated. Thanks!
>>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160518/e645c782/attachment.html