[Freeswitch-users] Disable annexb for g729

[Michael Jerris]

<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 May 17, 2016, at 2:37 PM, Michael Jerris <mike at jerris.com> wrote:
> 
> Interesting, https://tools.ietf.org/html/rfc7261 <https://tools.ietf.org/html/rfc7261> says annexb=yes is default if omitted.  Brian West, did you know that?
> 
> 
> 
>> On May 17, 2016, at 10:57 AM, Mike Rice <mrice0118 at gmail.com <mailto: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 <sip:XXXXXXXXXX at 10.10.X.X>>;reason=unavailable]
>> EXECUTE sofia/internal/1000 at 10.10.X.X <mailto: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 <mailto: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 <mailto:sofia/internal/1000 at 10.10.x.x> EXPORTING[export_vars] [sip_h_Diversion]=[<sip:XXXXXXXXXX at 10.10.X.X <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!
>> _________________________________________________________________________
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