[Freeswitch-users] mod_opus codec name.

Mon Nov 12 14:27:28 MSK 2012

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 at 107.22.240.239 SIP/2.0.

Call-ID: f3a55676d1f742c782a8eb400049d626 at 0:0:0:0:0:0:0:0.

CSeq: 2 INVITE.

From: "Stephen Dame" <sip:1000 at 107.22.240.239>;tag=771781e0.

To: <sip:5000 at 107.22.240.239>.

Max-Forwards: 70.

Contact: "Stephen Dame"
<sip:1000 at 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 at 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 at 107.22.240.239>;tag=771781e0.

To: <sip:5000 at 107.22.240.239>.

Call-ID: f3a55676d1f742c782a8eb400049d626 at 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 at 107.22.240.239>;tag=771781e0.

To: <sip:5000 at 107.22.240.239>;tag=pDHc0ma2Q58cr.

Call-ID: f3a55676d1f742c782a8eb400049d626 at 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 at 107.22.240.239>;party=calling;privacy=off;screen=no.

.

.

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at 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 at 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 at 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 at 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 at 10
7.22.240.239) Callstate Change DOWN -> HANGUP

2012-11-10 18:56:32.159321 [NOTICE] sofia.c:5743 Hangup
sofia/internal/1000 at 107.                                         22.240.239
[CS_NEW] [INCOMPATIBLE_DESTINATION]

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at 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 at 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 at 16000h@20i,speex at 8000h@20i,G7221 at 32000h,
G7221 at 16000h,G722,PCMU,PCMA,GSM" />

<X-PRE-PROCESS cmd="set"
data="outbound_codec_prefs=speex at 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

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