Hack for Zoiper speex codec payload problem
Hi All, I have problem with getting Zoiper with enabled codec speex/8000 working with FS. I saw some discussions about this problem before, but none of them ended with working solution. Maybe these days some of you can share some hints how to hack this Problem is that Zoiper sends SDP payload type 110 for speex/8000, but FS expects other type for that codec. I tried to rewrite SDP on OpenSIPS proxy that sits between end users and FS box, and then later rewrite replies with SDP payload generated by FS to 110 code, but calls always end-up with codec negotiation failure And yes, I know that zoiper is broken, please don't paste RFC here :) It gets very popular on mobile devices and I'm forced to provide support for it, even if its broken
Android or iOS? On Friday, October 17, 2014, Mateusz Bartczak <netcentrica@gmail.com> wrote:
Hi All,
I have problem with getting Zoiper with enabled codec speex/8000 working with FS. I saw some discussions about this problem before, but none of them ended with working solution.
Maybe these days some of you can share some hints how to hack this
Problem is that Zoiper sends SDP payload type 110 for speex/8000, but FS expects other type for that codec.
I tried to rewrite SDP on OpenSIPS proxy that sits between end users and FS box, and then later rewrite replies with SDP payload generated by FS to 110 code, but calls always end-up with codec negotiation failure
And yes, I know that zoiper is broken, please don't paste RFC here :)
It gets very popular on mobile devices and I'm forced to provide support for it, even if its broken
-- *Brian West* brian@freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
Android, but the same problem is with Windows client 2014-10-17 12:35 GMT+02:00 Brian West <brian@freeswitch.org>:
Android or iOS?
On Friday, October 17, 2014, Mateusz Bartczak <netcentrica@gmail.com> wrote:
Hi All,
I have problem with getting Zoiper with enabled codec speex/8000 working with FS. I saw some discussions about this problem before, but none of them ended with working solution.
Maybe these days some of you can share some hints how to hack this
Problem is that Zoiper sends SDP payload type 110 for speex/8000, but FS expects other type for that codec.
I tried to rewrite SDP on OpenSIPS proxy that sits between end users and FS box, and then later rewrite replies with SDP payload generated by FS to 110 code, but calls always end-up with codec negotiation failure
And yes, I know that zoiper is broken, please don't paste RFC here :)
It gets very popular on mobile devices and I'm forced to provide support for it, even if its broken
--
*Brian West* brian@freeswitch.org
*Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com
*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
_________________________________________________________________________ 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
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Since the codec is in the dynamic range the payload number is of little to no value the codec name is then used to compare. So could you provide some details of what you think is wrong? On Fri, Oct 17, 2014 at 7:34 AM, Mateusz Bartczak <netcentrica@gmail.com> wrote:
Android, but the same problem is with Windows client
2014-10-17 12:35 GMT+02:00 Brian West <brian@freeswitch.org>:
Android or iOS?
On Friday, October 17, 2014, Mateusz Bartczak <netcentrica@gmail.com> wrote:
Hi All,
I have problem with getting Zoiper with enabled codec speex/8000 working with FS. I saw some discussions about this problem before, but none of them ended with working solution.
Maybe these days some of you can share some hints how to hack this
Problem is that Zoiper sends SDP payload type 110 for speex/8000, but FS expects other type for that codec.
I tried to rewrite SDP on OpenSIPS proxy that sits between end users and FS box, and then later rewrite replies with SDP payload generated by FS to 110 code, but calls always end-up with codec negotiation failure
And yes, I know that zoiper is broken, please don't paste RFC here :)
It gets very popular on mobile devices and I'm forced to provide support for it, even if its broken
--
*Brian West* brian@freeswitch.org
*Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com
*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
_________________________________________________________________________ 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-powered IP PBX: The CudaTel Communication Server http://www.cudatel.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
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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-- *Brian West* brian@freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
Zoiper sends 101 code, which seems to be not recognized by FS absolute_codec_string is set to speex@8000h@20i,SILK@8000h @20i,G729,GSM,SILK@12000h@20i,PCMA,PCMU Here are parts from logs: 2014-11-15 00:01:26.118326 [DEBUG] sofia.c:5860 Remote SDP: v=0 o=Z 0 0 IN IP4 37.190.147.137 s=Z c=IN IP4 37.190.147.137 t=0 0 m=audio 64037 RTP/AVP 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5284 Audio Codec Compare [telephone-event:101:8000:20:0]/[PCMA:8:8000:20:64000] 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf send/recv payload to 101 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5284 Audio Codec Compare [telephone-event:101:8000:20:0]/[PCMU:0:8000:20:64000] 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf send/recv payload to 101 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5284 Audio Codec Compare [telephone-event:101:8000:20:0]/[SPEEX:99:32000:20:44000] 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf send/recv payload to 101 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5284 Audio Codec Compare [telephone-event:101:8000:20:0]/[SPEEX:99:16000:20:42200] 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf send/recv payload to 101 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5284 Audio Codec Compare [telephone-event:101:8000:20:0]/[SPEEX:99:8000:20:24600] 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf send/recv payload to 101 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5284 Audio Codec Compare [telephone-event:101:8000:20:0]/[SILK:118:12000:20:25000] 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf send/recv payload to 101 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5284 Audio Codec Compare [telephone-event:101:8000:20:0]/[SILK:117:8000:20:20000] 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf send/recv payload to 101 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5284 Audio Codec Compare [telephone-event:101:8000:20:0]/[G729:18:8000:20:8000] 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf send/recv payload to 101 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5284 Audio Codec Compare [telephone-event:101:8000:20:0]/[GSM:3:8000:20:13200] 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf send/recv payload to 101 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf send/recv payload to 101 2014-11-15 00:01:26.118326 [NOTICE] sofia.c:6185 Hangup sofia/external/ 964938@80.72.35.123 [CS_NEW] [INCOMPATIBLE_DESTINATION] Still no success 2014-10-21 14:29 GMT+02:00 Brian West <brian@freeswitch.org>:
Since the codec is in the dynamic range the payload number is of little to no value the codec name is then used to compare. So could you provide some details of what you think is wrong?
On Fri, Oct 17, 2014 at 7:34 AM, Mateusz Bartczak <netcentrica@gmail.com> wrote:
Android, but the same problem is with Windows client
2014-10-17 12:35 GMT+02:00 Brian West <brian@freeswitch.org>:
Android or iOS?
On Friday, October 17, 2014, Mateusz Bartczak <netcentrica@gmail.com> wrote:
Hi All,
I have problem with getting Zoiper with enabled codec speex/8000 working with FS. I saw some discussions about this problem before, but none of them ended with working solution.
Maybe these days some of you can share some hints how to hack this
Problem is that Zoiper sends SDP payload type 110 for speex/8000, but FS expects other type for that codec.
I tried to rewrite SDP on OpenSIPS proxy that sits between end users and FS box, and then later rewrite replies with SDP payload generated by FS to 110 code, but calls always end-up with codec negotiation failure
And yes, I know that zoiper is broken, please don't paste RFC here :)
It gets very popular on mobile devices and I'm forced to provide support for it, even if its broken
--
*Brian West* brian@freeswitch.org
*Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com
*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
_________________________________________________________________________ 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-powered IP PBX: The CudaTel Communication Server http://www.cudatel.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
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
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--
*Brian West* brian@freeswitch.org
*Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com
*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
_________________________________________________________________________ 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-powered IP PBX: The CudaTel Communication Server http://www.cudatel.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
Well considering 101 is telephony-event, Zoiper would appear to be broken. On Fri, Nov 14, 2014 at 5:07 PM, Mateusz Bartczak <netcentrica@gmail.com> wrote:
Zoiper sends 101 code, which seems to be not recognized by FS
absolute_codec_string is set to speex@8000h@20i,SILK@8000h @20i,G729,GSM,SILK@12000h@20i,PCMA,PCMU
Here are parts from logs:
2014-11-15 00:01:26.118326 [DEBUG] sofia.c:5860 Remote SDP: v=0 o=Z 0 0 IN IP4 37.190.147.137 s=Z c=IN IP4 37.190.147.137 t=0 0 m=audio 64037 RTP/AVP 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15
2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5284 Audio Codec Compare [telephone-event:101:8000:20:0]/[PCMA:8:8000:20:64000] 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf send/recv payload to 101 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5284 Audio Codec Compare [telephone-event:101:8000:20:0]/[PCMU:0:8000:20:64000] 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf send/recv payload to 101 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5284 Audio Codec Compare [telephone-event:101:8000:20:0]/[SPEEX:99:32000:20:44000] 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf send/recv payload to 101 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5284 Audio Codec Compare [telephone-event:101:8000:20:0]/[SPEEX:99:16000:20:42200] 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf send/recv payload to 101 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5284 Audio Codec Compare [telephone-event:101:8000:20:0]/[SPEEX:99:8000:20:24600] 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf send/recv payload to 101 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5284 Audio Codec Compare [telephone-event:101:8000:20:0]/[SILK:118:12000:20:25000] 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf send/recv payload to 101 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5284 Audio Codec Compare [telephone-event:101:8000:20:0]/[SILK:117:8000:20:20000] 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf send/recv payload to 101 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5284 Audio Codec Compare [telephone-event:101:8000:20:0]/[G729:18:8000:20:8000] 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf send/recv payload to 101 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5284 Audio Codec Compare [telephone-event:101:8000:20:0]/[GSM:3:8000:20:13200] 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf send/recv payload to 101 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf send/recv payload to 101 2014-11-15 00:01:26.118326 [NOTICE] sofia.c:6185 Hangup sofia/external/ 964938@80.72.35.123 [CS_NEW] [INCOMPATIBLE_DESTINATION]
Still no success
2014-10-21 14:29 GMT+02:00 Brian West <brian@freeswitch.org>:
Since the codec is in the dynamic range the payload number is of little to no value the codec name is then used to compare. So could you provide some details of what you think is wrong?
On Fri, Oct 17, 2014 at 7:34 AM, Mateusz Bartczak <netcentrica@gmail.com> wrote:
Android, but the same problem is with Windows client
2014-10-17 12:35 GMT+02:00 Brian West <brian@freeswitch.org>:
Android or iOS?
On Friday, October 17, 2014, Mateusz Bartczak <netcentrica@gmail.com> wrote:
Hi All,
I have problem with getting Zoiper with enabled codec speex/8000 working with FS. I saw some discussions about this problem before, but none of them ended with working solution.
Maybe these days some of you can share some hints how to hack this
Problem is that Zoiper sends SDP payload type 110 for speex/8000, but FS expects other type for that codec.
I tried to rewrite SDP on OpenSIPS proxy that sits between end users and FS box, and then later rewrite replies with SDP payload generated by FS to 110 code, but calls always end-up with codec negotiation failure
And yes, I know that zoiper is broken, please don't paste RFC here :)
It gets very popular on mobile devices and I'm forced to provide support for it, even if its broken
--
*Brian West* brian@freeswitch.org
*Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com
*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
_________________________________________________________________________ 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-powered IP PBX: The CudaTel Communication Server http://www.cudatel.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
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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--
*Brian West* brian@freeswitch.org
*Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com
*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
_________________________________________________________________________ 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-powered IP PBX: The CudaTel Communication Server http://www.cudatel.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
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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-- *Brian West* brian@freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
That¹s just broken in general... The RTP map there is saying that codec 101 is telephone-event/8000 which is specifically RFC2833 DTMF... There is no audio codec in there... In this case it appears no actual audio codecs were offered On 11/14/14 5:07 PM, "Mateusz Bartczak" <netcentrica@gmail.com> wrote:
Zoiper sends 101 code, which seems to be not recognized by FS
absolute_codec_string is set to speex@8000h@20i,SILK@8000h@20i,G729,GSM,SILK@12000h@20i,PCMA,PCMU
Here are parts from logs:
2014-11-15 00:01:26.118326 [DEBUG] sofia.c:5860 Remote SDP: v=0 o=Z 0 0 IN IP4 37.190.147.137 s=Z c=IN IP4 37.190.147.137 t=0 0 m=audio 64037 RTP/AVP 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15
2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5284 Audio Codec Compare [telephone-event:101:8000:20:0]/[PCMA:8:8000:20:64000] 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf send/recv payload to 101 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5284 Audio Codec Compare [telephone-event:101:8000:20:0]/[PCMU:0:8000:20:64000] 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf send/recv payload to 101 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5284 Audio Codec Compare [telephone-event:101:8000:20:0]/[SPEEX:99:32000:20:44000] 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf send/recv payload to 101 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5284 Audio Codec Compare [telephone-event:101:8000:20:0]/[SPEEX:99:16000:20:42200] 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf send/recv payload to 101 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5284 Audio Codec Compare [telephone-event:101:8000:20:0]/[SPEEX:99:8000:20:24600] 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf send/recv payload to 101 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5284 Audio Codec Compare [telephone-event:101:8000:20:0]/[SILK:118:12000:20:25000] 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf send/recv payload to 101 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5284 Audio Codec Compare [telephone-event:101:8000:20:0]/[SILK:117:8000:20:20000] 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf send/recv payload to 101 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5284 Audio Codec Compare [telephone-event:101:8000:20:0]/[G729:18:8000:20:8000] 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf send/recv payload to 101 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5284 Audio Codec Compare [telephone-event:101:8000:20:0]/[GSM:3:8000:20:13200] 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf send/recv payload to 101 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf send/recv payload to 101 2014-11-15 00:01:26.118326 [NOTICE] sofia.c:6185 Hangup sofia/external/964938@80.72.35.123 [CS_NEW] [INCOMPATIBLE_DESTINATION]
Still no success
2014-10-21 14:29 GMT+02:00 Brian West <brian@freeswitch.org>:
Since the codec is in the dynamic range the payload number is of little to no value the codec name is then used to compare. So could you provide some details of what you think is wrong?
On Fri, Oct 17, 2014 at 7:34 AM, Mateusz Bartczak <netcentrica@gmail.com> wrote:
Android, but the same problem is with Windows client
2014-10-17 12:35 GMT+02:00 Brian West <brian@freeswitch.org>:
Android or iOS?
On Friday, October 17, 2014, Mateusz Bartczak <netcentrica@gmail.com> wrote:
Hi All,
I have problem with getting Zoiper with enabled codec speex/8000 working with FS. I saw some discussions about this problem before, but none of them ended with working solution.
Maybe these days some of you can share some hints how to hack this
Problem is that Zoiper sends SDP payload type 110 for speex/8000, but FS expects other type for that codec.
I tried to rewrite SDP on OpenSIPS proxy that sits between end users and FS box, and then later rewrite replies with SDP payload generated by FS to 110 code, but calls always end-up with codec negotiation failure
And yes, I know that zoiper is broken, please don't paste RFC here :)
It gets very popular on mobile devices and I'm forced to provide support for it, even if its broken
-- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH
participants (3)
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Brian West
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Ken Rice
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Mateusz Bartczak