Hi, We are facing an issue with a=fmtp:18 annexb=no. Here is the call flow, Cisco CUCM -> Cisco Cube -> FS Cisco is expecting in 200 OK , m=audio 28820 RTP/AVP 18 101 13 a=rtpmap:18 G729/8000 * a=fmtp:18 annexb=no* a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 But fs is generating without *a=fmtp:18 annexb=no* m=audio 28820 RTP/AVP 18 101 13 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 Because of that calls are getting fail. I am not getting where should we change setting to include that line. Log, http://pastebin.com/LDHEmCNv Please help me to resolve that issue. Thanks in advance. ----- Thanks, Sam -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Issue-with-a-fmtp-18-annexb-no... Sent from the freeswitch-users mailing list archive at Nabble.com.
application="export" data="sip_append_audio_sdp=a=fmtp:18 annexb=no" On Thursday 13 February 2014 07:09:16 Sam wrote:
Hi,
We are facing an issue with a=fmtp:18 annexb=no.
Here is the call flow, Cisco CUCM -> Cisco Cube -> FS
Cisco is expecting in 200 OK ,
m=audio 28820 RTP/AVP 18 101 13 a=rtpmap:18 G729/8000 * a=fmtp:18 annexb=no* a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20
But fs is generating without *a=fmtp:18 annexb=no* m=audio 28820 RTP/AVP 18 101 13 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20
Because of that calls are getting fail.
I am not getting where should we change setting to include that line.
Log, http://pastebin.com/LDHEmCNv
Please help me to resolve that issue.
Thanks in advance.
----- Thanks, Sam -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Issue-with-a-fmtp-18-annexb-n o-tp7596060.html Sent from the freeswitch-users mailing list archive at Nabble.com.
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Nevermind, my suggestion will work for bridging calls, but no for answering. On Thursday 13 February 2014 10:43:45 Sergey Okhapkin wrote:
application="export" data="sip_append_audio_sdp=a=fmtp:18 annexb=no"
On Thursday 13 February 2014 07:09:16 Sam wrote:
Hi,
We are facing an issue with a=fmtp:18 annexb=no.
Here is the call flow, Cisco CUCM -> Cisco Cube -> FS
Cisco is expecting in 200 OK ,
m=audio 28820 RTP/AVP 18 101 13 a=rtpmap:18 G729/8000 * a=fmtp:18 annexb=no* a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20
But fs is generating without *a=fmtp:18 annexb=no*
m=audio 28820 RTP/AVP 18 101 13 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20
Because of that calls are getting fail.
I am not getting where should we change setting to include that line.
Log, http://pastebin.com/LDHEmCNv
Please help me to resolve that issue.
Thanks in advance.
----- Thanks, Sam -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Issue-with-a-fmtp-18-annexb-> > n o-tp7596060.html Sent from the freeswitch-users mailing list archive at Nabble.com.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.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
Right I need to set that for A leg INVITE in 200 OK. On Thu, Feb 13, 2014 at 9:20 PM, Sergey Okhapkin <sos@sokhapkin.dyndns.org>wrote:
Nevermind, my suggestion will work for bridging calls, but no for answering.
On Thursday 13 February 2014 10:43:45 Sergey Okhapkin wrote:
application="export" data="sip_append_audio_sdp=a=fmtp:18 annexb=no"
On Thursday 13 February 2014 07:09:16 Sam wrote:
Hi,
We are facing an issue with a=fmtp:18 annexb=no.
Here is the call flow, Cisco CUCM -> Cisco Cube -> FS
Cisco is expecting in 200 OK ,
m=audio 28820 RTP/AVP 18 101 13 a=rtpmap:18 G729/8000 * a=fmtp:18 annexb=no* a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20
But fs is generating without *a=fmtp:18 annexb=no*
m=audio 28820 RTP/AVP 18 101 13 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20
Because of that calls are getting fail.
I am not getting where should we change setting to include that line.
Log, http://pastebin.com/LDHEmCNv
Please help me to resolve that issue.
Thanks in advance.
----- Thanks, Sam -- View this message in context:
http://freeswitch-users.2379917.n2.nabble.com/Issue-with-a-fmtp-18-annexb-> n
o-tp7596060.html Sent from the freeswitch-users mailing list archive at Nabble.com.
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.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
Cisco requires Annexb=no in 200 ok A workaround for this is late offer with “rtp_append_audio_sdp=a=fmtp:18 annexb=no” but if we do that we get “No SDP in INVITE and 3pcc not enabled, hanging up.” (Early Offer isn’t a great solution for this as it causes other problems) So, is there any other way of get around either of these issues? First prize is Annexb=no Please let me know. ----- Thanks, Sam -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Issue-with-a-fmtp-18-annexb-no... Sent from the freeswitch-users mailing list archive at Nabble.com.
I have the exact same problem and would love a solution. The early offer causes another problem to arise with "fake ring" Surely there's a way to either send annexb or ignore not getting sdp. I think either would resolve this. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Issue-with-a-fmtp-18-annexb-no... Sent from the freeswitch-users mailing list archive at Nabble.com.
participants (4)
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jacoj
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Sam
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Samir Doshi
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Sergey Okhapkin