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/$1"/> inbound-late-negotiation is set to true. The logs show that it is exporting: [DEBUG] switch_channel.c:1267 EXPORT (export_vars) [sip_h_Diversion]=[<sip:XXXXXXXXXX@10.10.X.X>;reason=unavailable] EXECUTE sofia/internal/1000@10.10.X.X bridge(sofia/gateway/lo7f/XXXXXXXXXX) [DEBUG] switch_channel.c:1221 sofia/internal/1000@10.10.X.X EXPORTING[export_vars] [sip_append_audio_sdp]=[a=fmtp:18 annexb=no] to event [DEBUG] switch_channel.c:1221 sofia/internal/1000@10.10.X.X EXPORTING[export_vars] [sip_h_Diversion]=[<sip:XXXXXXXXXX@10.10.X.X>;reason=unavailable] to event The SDP does not reflect: Local SDP: v=0 o=FreeSWITCH IN IP4 10.10.X.X s=FreeSWITCH c=IN IP4 10.10.X.X t=0 0 m=audio 20022 RTP/AVP 18 0 8 101 13 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv Any help would be greatly appreciated. Thanks!
Interesting, https://tools.ietf.org/html/rfc7261 <https://tools.ietf.org/html/rfc7261> says annexb=yes is default if omitted. Brian West, did you know that?
On May 17, 2016, at 10:57 AM, Mike Rice <mrice0118@gmail.com> wrote:
We have a carrier the mandates that annex b be disabled on the invites to them. I have added the following to the default dialplan but nothing seems to change the invite on the B leg to the carrier.
<extension name="disable-annexB" continue="true"> <condition field="${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
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
<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@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@gmail.com <mailto: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 <sip:XXXXXXXXXX@10.10.X.X>>;reason=unavailable] EXECUTE sofia/internal/1000@10.10.X.X <mailto:sofia/internal/1000@10.10.x.x> bridge(sofia/gateway/lo7f/XXXXXXXXXX) [DEBUG] switch_channel.c:1221 sofia/internal/1000@10.10.X.X <mailto: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 <mailto:sofia/internal/1000@10.10.x.x> EXPORTING[export_vars] [sip_h_Diversion]=[<sip:XXXXXXXXXX@10.10.X.X <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 <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Thanks Michael! That did the trick. Did I miss that somewhere? From: Michael Jerris <mike@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@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/$1"/>
inbound-late-negotiation is set to true. The logs show that it is exporting:
[DEBUG] switch_channel.c:1267 EXPORT (export_vars) [sip_h_Diversion]=[<sip:XXXXXXXXXX@10.10.X.X>;reason=unavailable] EXECUTE sofia/internal/1000@10.10.X.X bridge(sofia/gateway/lo7f/XXXXXXXXXX) [DEBUG] switch_channel.c:1221 sofia/internal/1000@10.10.X.X EXPORTING[export_vars] [sip_append_audio_sdp]=[a=fmtp:18 annexb=no] to event [DEBUG] switch_channel.c:1221 sofia/internal/1000@10.10.X.X EXPORTING[export_vars] [sip_h_Diversion]=[<sip:XXXXXXXXXX@10.10.X.X>;reason=unavailable] to event
The SDP does not reflect:
Local SDP: v=0 o=FreeSWITCH IN IP4 10.10.X.X s=FreeSWITCH c=IN IP4 10.10.X.X t=0 0 m=audio 20022 RTP/AVP 18 0 8 101 13 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv
Any help would be greatly appreciated. Thanks!
It was spread about pretty widely when we went to 1.4 but thats a couple years ago now. It should be documented on the wiki but may have fallen off during the transition to confluence as 1.2 is quite old. Where did you see docs for the sip_ var?
On May 17, 2016, at 3:51 PM, Mike Rice <mrice0118@gmail.com> wrote:
Thanks Michael! That did the trick. Did I miss that somewhere?
From: Michael Jerris <mike@jerris.com <mailto:mike@jerris.com>>
<action application="export" data="rtp_append_audio_sdp=a=fmtp:18 annexb=no"/>
between 1.2 and 1.4 the var got renamed from sip_ to rtp_
On Tue, May 17, 2016 at 10:57 AM, Mike Rice <mrice0118@gmail.com <mailto:mrice0118@gmail.com>> wrote: We have a carrier the mandates that annex b be disabled on the invites to them. I have added the following to the default dialplan but nothing seems to change the invite on the B leg to the carrier.
<extension name="disable-annexB" continue="true"> <condition field="${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
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
I got it from the following URL. https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation ---------- Forwarded message ---------- From: Michael Jerris <mike@jerris.com> To: FreeSWITCH Users Help <freeswitch-users@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@gmail.com> wrote:
Thanks Michael! That did the trick. Did I miss that somewhere?
From: Michael Jerris <mike@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@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!
fixed, thanks.
On May 18, 2016, at 12:20 PM, Mike Rice <mrice0118@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@jerris.com <mailto:mike@jerris.com>> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org <mailto:freeswitch-users@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@gmail.com <mailto:mrice0118@gmail.com>> wrote: Thanks Michael! That did the trick. Did I miss that somewhere?
From: Michael Jerris <mike@jerris.com <mailto:mike@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@gmail.com <mailto: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!
participants (2)
-
Michael Jerris
-
Mike Rice