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@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@10.10.X.X>;reason=unavailable] EXECUTE sofia/internal/1000@10.10.X.X bridge(sofia/gateway/lo7f/XXXXXXXXXX) [DEBUG] switch_channel.c:1221 sofia/internal/1000@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@10.10.X.X EXPORTING[export_vars] [sip_h_Diversion]=[<sip:XXXXXXXXXX@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! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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