Anton, thanks for info, I get call not acceptable here. So no codec seems to be matching. is OPUS the right value, it never shows when its searching. c=IN IP4 192.168.99.198 t=0 0 m=audio 5032 RTP/AVP 96 a=rtpmap:96 opus/48000 a=extmap:1 urn:ietf:params:rtp-hdrext:csrc-audio-level a=zrtp-hash:1.10 6e63145d4d42de5a83db9967b61e247360ccf093e5d3299eb3605b976901f71 e m=video 5034 RTP/AVP 97 99 a=rtpmap:97 H264/90000 a=fmtp:97 profile-level-id=4DE01f;packetization-mode=1 a=rtpmap:99 H264/90000 a=fmtp:99 profile-level-id=4DE01f a=recvonly a=imageattr:97 send [x=[0-640],y=[0-480]] recv [x=[0-1366],y=[0-768]] a=imageattr:99 send [x=[0-640],y=[0-480]] recv [x=[0-1366],y=[0-768]] a=zrtp-hash:1.10 14c0fd7b5b9c4386a10aa646894fdf8f03710de838f71e8241fca33b79a6592 9 2012-11-10 18:56:32.159321 [DEBUG] switch_core_state_machine.c:362 (sofia/intern al/1000@107.22.240.239) Running State Change CS_NEW 2012-11-10 18:56:32.159321 [DEBUG] switch_core_state_machine.c:380 (sofia/intern al/1000@107.22.240.239) State NEW 2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [opus:9 6:48000:20:0]/[G726-24:123:8000:20:24000] 2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [opus:9 6:48000:20:0]/[G729:18:8000:20:8000] 2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [opus:9 6:48000:20:0]/[SPEEX:99:32000:20:44000] 2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [opus:9 6:48000:20:0]/[SPEEX:99:16000:20:42200] 2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [opus:9 6:48000:20:0]/[SPEEX:99:8000:20:24600] 2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [opus:9 6:48000:20:0]/[G7221:115:32000:20:48000] 2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [opus:9 6:48000:20:0]/[G7221:107:16000:20:32000] 2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [opus:9 6:48000:20:0]/[G722:9:8000:20:64000] 2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [opus:9 6:48000:20:0]/[PCMU:0:8000:20:64000] 2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [opus:9 6:48000:20:0]/[PCMA:8:8000:20:64000] 2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [opus:9 6:48000:20:0]/[GSM:3:8000:20:13200] 2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4930 No 2833 in SDP. Disable 28 33 dtmf and switch to INFO 2012-11-10 18:56:32.159321 [DEBUG] switch_channel.c:2846 (sofia/internal/1000@10 7.22.240.239) Callstate Change DOWN -> HANGUP 2012-11-10 18:56:32.159321 [NOTICE] sofia.c:5743 Hangup sofia/internal/1000@107. 22.240.239 [CS_NEW] [INCOMPATIBLE_DESTINATION] From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Anton Kvashenkin Sent: Saturday, November 10, 2012 3:34 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] mod_opus codec name. As we see in sdp - it's definitely works, but I'm not shure about jitsi. When I was testing jitsi nightly build for OPUS support it was not working at all. 2012/11/10 Stephen Dame <sdame@207me.com> I have mod_opus loaded. Trying to find the documentation as to what the name of the CODEC is for including it in codec preferences. I tried OPUS, but that doesn't seem to work. <X-PRE-PROCESS cmd="set" data="global_codec_prefs=OPUS,speex@16000h@20i,speex@8000h@20i,G7221@32000h, G7221@16000h,G722,PCMU,PCMA,GSM" /> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=speex@16000h@20i,PCMU,PCMA,GSM" /> Trying to get Jitsi to negotiate 2012-11-10 12:54:22.039330 [DEBUG] sofia.c:5475 Remote SDP: v=0 o=1000 0 0 IN IP4 192.168.99.198 s=- c=IN IP4 192.168.99.198 t=0 0 m=audio 5020 RTP/AVP 96 a=rtpmap:96 opus/48000 a=extmap:1 urn:ietf:params:rtp-hdrext:csrc-audio-level a=zrtp-hash:1.10 29740c437bd6d867c08a8854b65368c5052c70ec0b463a06f748062657529239 thanks in advance, I googled the list, and docs don't seem to find the answer. Regards, Stephen _________________________________________________________________________ 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

Anton, I have reloaded and restarted FS. And made sure mod_opus is loaded. I still see no documentation in forums or wiki, as to what name should be used for opus in global codec preferences. I get 488, which looks like freeswitch is never looking to compare it against the preference list. Thanks Stephen T 2012/11/12 06:20:38.776257 67.253.29.100:42778 -> 10.126.171.197:5080 [AP] INVITE sip:5000@107.22.240.239 SIP/2.0. Call-ID: f3a55676d1f742c782a8eb400049d626@0:0:0:0:0:0:0:0. CSeq: 2 INVITE. From: "Stephen Dame" <sip:1000@107.22.240.239>;tag=771781e0. To: <sip:5000@107.22.240.239>. Max-Forwards: 70. Contact: "Stephen Dame" <sip:1000@192.168.99.198:50441;transport=tcp;registering_acc=107_22_240_239> . User-Agent: Jitsi1.1.4310.10045Windows 7. Content-Type: application/sdp. Via: SIP/2.0/TCP 192.168.99.198:50441;branch=z9hG4bK-333236-9a42b9b6a906d0549253b69d9eb87431. Proxy-Authorization: Digest username="1000",realm="107.22.240.239",nonce="9afe2f18-8924-41be-9cab-82c371 bc7529",uri="sip:5000@107.22.240.239",response="f93d24497a79f51d81bee9a3a851 81bf",algorithm=MD5,qop=auth,cnonce="xyz",nc=00000001. Content-Length: 668. . v=0. o=1000 0 0 IN IP4 192.168.99.198. s=-. c=IN IP4 192.168.99.198. t=0 0. m=audio 5060 RTP/AVP 96. a=rtpmap:96 opus/48000. a=extmap:1 urn:ietf:params:rtp-hdrext:csrc-audio-level. a=zrtp-hash:1.10 fa977be972f6bf8a035e3519ba22b89dfbf191750ec9c3d027ce9cc73c7da199. m=video 5062 RTP/AVP 97 99. a=recvonly. a=rtpmap:97 H264/90000. a=fmtp:97 profile-level-id=4DE01f;packetization-mode=1. a=imageattr:97 send [x=[0-640],y=[0-480]] recv [x=[0-1366],y=[0-768]]. a=rtpmap:99 H264/90000. a=fmtp:99 profile-level-id=4DE01f. a=imageattr:99 send [x=[0-640],y=[0-480]] recv [x=[0-1366],y=[0-768]]. a=zrtp-hash:1.10 711982f565cb5da0fe3added9c7c50e046df17a0d44b463371900ab295bc972f. T 2012/11/12 06:20:38.776481 10.126.171.197:5080 -> 67.253.29.100:42778 [AP] SIP/2.0 100 Trying. Via: SIP/2.0/TCP 192.168.99.198:50441;branch=z9hG4bK-333236-9a42b9b6a906d0549253b69d9eb87431; received=67.253.29.100;rport=42778. From: "Stephen Dame" <sip:1000@107.22.240.239>;tag=771781e0. To: <sip:5000@107.22.240.239>. Call-ID: f3a55676d1f742c782a8eb400049d626@0:0:0:0:0:0:0:0. CSeq: 2 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1ddd29b 2011-12-18 12-08-17 -0500. Content-Length: 0. . T 2012/11/12 06:20:38.783929 10.126.171.197:5080 -> 67.253.29.100:42778 [AP] SIP/2.0 488 Not Acceptable Here. Via: SIP/2.0/TCP 192.168.99.198:50441;branch=z9hG4bK-333236-9a42b9b6a906d0549253b69d9eb87431; received=67.253.29.100;rport=42778. From: "Stephen Dame" <sip:1000@107.22.240.239>;tag=771781e0. To: <sip:5000@107.22.240.239>;tag=pDHc0ma2Q58cr. Call-ID: f3a55676d1f742c782a8eb400049d626@0:0:0:0:0:0:0:0. CSeq: 2 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1ddd29b 2011-12-18 12-08-17 -0500. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Content-Length: 0. Remote-Party-ID: "5000" <sip:5000@107.22.240.239>;party=calling;privacy=off;screen=no. . . From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Anton Kvashenkin Sent: Sunday, November 11, 2012 1:26 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] mod_opus codec name. Don't forget to make reloadxml (F6). Can you paste full debug log with <sofia profile internla siptrace on> or ngrep -d <your iface> -qt -W byline port 5060 and host <which making call> 2012/11/11 Stephen Dame <sdame@207me.com> Anton, thanks for info, I get call not acceptable here. So no codec seems to be matching. is OPUS the right value, it never shows when its searching. c=IN IP4 192.168.99.198 t=0 0 m=audio 5032 RTP/AVP 96 a=rtpmap:96 opus/48000 a=extmap:1 urn:ietf:params:rtp-hdrext:csrc-audio-level a=zrtp-hash:1.10 6e63145d4d42de5a83db9967b61e247360ccf093e5d3299eb3605b976901f71 e m=video 5034 RTP/AVP 97 99 a=rtpmap:97 H264/90000 a=fmtp:97 profile-level-id=4DE01f;packetization-mode=1 a=rtpmap:99 H264/90000 a=fmtp:99 profile-level-id=4DE01f a=recvonly a=imageattr:97 send [x=[0-640],y=[0-480]] recv [x=[0-1366],y=[0-768]] a=imageattr:99 send [x=[0-640],y=[0-480]] recv [x=[0-1366],y=[0-768]] a=zrtp-hash:1.10 14c0fd7b5b9c4386a10aa646894fdf8f03710de838f71e8241fca33b79a6592 9 2012-11-10 18:56:32.159321 [DEBUG] switch_core_state_machine.c:362 (sofia/intern al/1000@107.22.240.239) Running State Change CS_NEW 2012-11-10 18:56:32.159321 [DEBUG] switch_core_state_machine.c:380 (sofia/intern al/1000@107.22.240.239) State NEW 2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [opus:9 6:48000:20:0]/[G726-24:123:8000:20:24000] 2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [opus:9 6:48000:20:0]/[G729:18:8000:20:8000] 2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [opus:9 6:48000:20:0]/[SPEEX:99:32000:20:44000] 2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [opus:9 6:48000:20:0]/[SPEEX:99:16000:20:42200] 2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [opus:9 6:48000:20:0]/[SPEEX:99:8000:20:24600] 2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [opus:9 6:48000:20:0]/[G7221:115:32000:20:48000] 2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [opus:9 6:48000:20:0]/[G7221:107:16000:20:32000] 2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [opus:9 6:48000:20:0]/[G722:9:8000:20:64000] 2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [opus:9 6:48000:20:0]/[PCMU:0:8000:20:64000] 2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [opus:9 6:48000:20:0]/[PCMA:8:8000:20:64000] 2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [opus:9 6:48000:20:0]/[GSM:3:8000:20:13200] 2012-11-10 18:56:32.159321 [DEBUG] sofia_glue.c:4930 No 2833 in SDP. Disable 28 33 dtmf and switch to INFO 2012-11-10 18:56:32.159321 [DEBUG] switch_channel.c:2846 (sofia/internal/1000@10 7.22.240.239) Callstate Change DOWN -> HANGUP 2012-11-10 18:56:32.159321 [NOTICE] sofia.c:5743 Hangup sofia/internal/1000@107. 22.240.239 [CS_NEW] [INCOMPATIBLE_DESTINATION] From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Anton Kvashenkin Sent: Saturday, November 10, 2012 3:34 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] mod_opus codec name. As we see in sdp - it's definitely works, but I'm not shure about jitsi. When I was testing jitsi nightly build for OPUS support it was not working at all. 2012/11/10 Stephen Dame <sdame@207me.com> I have mod_opus loaded. Trying to find the documentation as to what the name of the CODEC is for including it in codec preferences. I tried OPUS, but that doesn't seem to work. <X-PRE-PROCESS cmd="set" data="global_codec_prefs=OPUS,speex@16000h@20i,speex@8000h@20i,G7221@32000h, G7221@16000h,G722,PCMU,PCMA,GSM" /> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=speex@16000h@20i,PCMU,PCMA,GSM" /> Trying to get Jitsi to negotiate 2012-11-10 12:54:22.039330 [DEBUG] sofia.c:5475 Remote SDP: v=0 o=1000 0 0 IN IP4 192.168.99.198 s=- c=IN IP4 192.168.99.198 t=0 0 m=audio 5020 RTP/AVP 96 a=rtpmap:96 opus/48000 a=extmap:1 urn:ietf:params:rtp-hdrext:csrc-audio-level a=zrtp-hash:1.10 29740c437bd6d867c08a8854b65368c5052c70ec0b463a06f748062657529239 thanks in advance, I googled the list, and docs don't seem to find the answer. Regards, Stephen _________________________________________________________________________ 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

Incorrect. If you have a version of the module from before the codec was released officially it would be opus-0.9.0. That is now discontinued so you would need to update to latest and use "opus" On Mon, Nov 12, 2012 at 9:29 AM, Frank Carmickle <frank@carmickle.com>wrote:

On Nov 12, 2012, at 6:27 AM, Stephen Dame <sdame@207me.com> wrote:

...

Anton, I have reloaded and restarted FS. And made sure mod_opus is loaded.

I still see no documentation in forums or wiki, as to what name should be used for opus in global codec preferences.

I get 488, which looks like freeswitch is never looking to compare it against the preference list.

It used to be Opus-0.9.0. I believe it is now Opus-1.0.1.

HTH --FC

_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com

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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900