[asterisk-bugs] [JIRA] (ASTERISK-23190) WebRTC (WSS + TLS) No Audio

Mon Jan 27 18:21:03 CST 2014

    [ https://issues.asterisk.org/jira/browse/ASTERISK-23190?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=214472#comment-214472 ] 

Jay Jideliov commented on ASTERISK-23190:
-----------------------------------------

This ticket was created in response to the comment in 21930 (https://issues.asterisk.org/jira/browse/ASTERISK-21930?focusedCommentId=212881&page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel#comment-212881).

The issue may not be related to the patch altogether, the information above was included only to depict the situation in full (which may help).

Moises himself has stated that this is an unrelated issue:
"It seems with Asterisk 11 branch I have no audio (even without my patch, so is not related)." (https://issues.asterisk.org/jira/browse/ASTERISK-21930?focusedCommentId=210859&page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel#comment-210859)

> WebRTC (WSS + TLS) No Audio
> ---------------------------
>
>                 Key: ASTERISK-23190
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-23190
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>    Affects Versions: 11.7.0
>            Reporter: Jay Jideliov
>
> This is a follow-up ticket that relates to asterisk patched as described here: 
> Patch 1:https://issues.asterisk.org/jira/browse/ASTERISK-22961
> Patch 2: https://issues.asterisk.org/jira/browse/ASTERISK-21930
> After getting SRTP and WSS to work, we are still experiencing the lack of sound. We believe that this is another bug that has to be fixed to obtain a fully working WebRTC secure deployment (WSS) functioning on Chrome/Firefox/Opera, finally pushing it to a normal (functioning) state.
> 1) We had a clean 11.7 to which we have added the patch from #22961
> 2) This got us to a working WebRTC on Chrome/Firefox over WS
> 3) Then, we have applied the patch from #21930 to get WSS to work
> 4) A call was established using SIPML5 over WSS, however, no sound was flowing
> The only error in console is: SetRemoteDescription failed: Called with an SDP without ice-ufrag and ice-pwd. (although we believe that this has nothing to do with the issue).
> The CLI gives us this:
>    -- Registered SIP 'xxx.device-1424' at xx.x.x.x:50560
>   == Using SIP RTP CoS mark 5
> [Jan 27 18:21:40] WARNING[29840][C-0000044e]: chan_sip.c:10496 process_sdp: Processed DTLS [TRUE]
>        > [INSERT INTO asterisk_db_cel (eventtype,eventtime,cid_name,cid_num,cid_ani,cid_rdnis,cid_dnid,exten,context,channame,appname,appdata,amaflags,accountcode,uniqueid,linkedid,peer,userdeftype,userfield) VALUES ('CHAN_START',{ts '2014-01-27 18:21:40'},'xxx.device-1424','xxx.device-1424','','','','888','incoming','SIP/xxx.device-1424-000000a7','','',3,'','1390864900.168','1390864900.168','','','')]
>     -- Executing [888 at incoming:1] Playback("SIP/xxxx.device-1424-000000a7", "demo-echotest") in new stack
>     --   >> Doing DTLS handshake as well...
>     --   >> [activate] check pending...
>        > [INSERT INTO asterisk_db_cel (eventtype,eventtime,cid_name,cid_num,cid_ani,cid_rdnis,cid_dnid,exten,context,channame,appname,appdata,amaflags,accountcode,uniqueid,linkedid,peer,userdeftype,userfield) VALUES ('ANSWER',{ts '2014-01-27 18:21:40'},'xxx.device-1424','xxx.device-1424','xxx.device-1424','','888','888','incoming','SIP/xxx.device-1424-000000a7','Playback','demo-echotest',3,'','1390864900.168','1390864900.168','','','')]
> Sent RTP packet to      x.x.x.x:64402 (via ICE) (type 00, seq 048496, ts 000160, len 4294967284)
>     -- <SIP/xxxx.device-1424-000000a7> Playing 'demo-echotest.gsm' (language 'en')
> Sent RTP packet to      x.x.x.x:64402 (via ICE) (type 00, seq 048497, ts 000320, len 4294967284)
> Sent RTP packet to      x.x.x.x:64402 (via ICE) (type 00, seq 048498, ts 000480, len 4294967284)
> Sent RTP packet to      x.x.x.x:64402 (via ICE) (type 00, seq 048499, ts 000640, len 4294967284)
> Sent RTP packet to      x.x.x.x:64402 (via ICE) (type 00, seq 048500, ts 000800, len 4294967284)
> Sent RTP packet to      x.x.x.x:64402 (via ICE) (type 00, seq 048501, ts 000960, len 4294967284)
> Sent RTP packet to      x.x.x.x:64402 (via ICE) (type 00, seq 048502, ts 001120, len 4294967284)
> Sent RTP packet to      x.x.x.x:64402 (via ICE) (type 00, seq 048503, ts 001280, len 4294967284)
> Sent RTP packet to      x.x.x.x:64402 (via ICE) (type 00, seq 048504, ts 001440, len 4294967284)
> Sent RTP packet to      x.x.x.x:64402 (via ICE) (type 00, seq 048505, ts 001600, len 4294967284)
> Sent RTP packet to      x.x.x.x:64402 (via ICE) (type 00, seq 048506, ts 001760, len 4294967284)
> Sent RTP packet to      x.x.x.x:64402 (via ICE) (type 00, seq 048507, ts 001920, len 4294967284)
> Sent RTP packet to      x.x.x.x:64402 (via ICE) (type 00, seq 048508, ts 002080, len 4294967284)
> Sent RTP packet to      x.x.x.x:64402 (via ICE) (type 00, seq 048509, ts 002240, len 4294967284)
> Sent RTP packet to      x.x.x.x:64402 (via ICE) (type 00, seq 048510, ts 002400, len 4294967284)
> Sent RTP packet to      x.x.x.x:64402 (via ICE) (type 00, seq 048511, ts 002560, len 4294967284)

--
This message is automatically generated by JIRA.
If you think it was sent incorrectly, please contact your JIRA administrators: https://issues.asterisk.org/jira/secure/ContactAdministrators!default.jspa
For more information on JIRA, see: http://www.atlassian.com/software/jira