[Freeswitch-users] Disable annexb for g729

[Michael Jerris]

fixed, thanks.

> On May 18, 2016, at 12:20 PM, Mike Rice <mrice0118 at gmail.com> wrote:
> 
> I got it from the following URL.
> 
> https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation <https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation>
> 
> ---------- Forwarded message ----------
> From: Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto: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 <mailto:mrice0118 at gmail.com>> wrote:
> Thanks Michael! That did the trick. Did I miss that somewhere?
> 
> From: Michael Jerris <mike at jerris.com <mailto: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 <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>;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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