[Freeswitch-users] Disable annexb for g729

Michael Jerris mike at jerris.com
Tue May 17 22:37:28 MSD 2016


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> 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!
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