| Summary: | ASTERISK-23190: WebRTC (WSS + TLS) No Audio | ||||
| Reporter: | Jay Jideliov (jideliov) | Labels: | |||
| Date Opened: | 2014-01-27 17:24:36.000-0600 | Date Closed: | 2014-01-27 18:09:49.000-0600 | ||
| Priority: | Major | Regression? | |||
| Status: | Closed/Complete | Components: | |||
| Versions: | 11.7.0 | Frequency of Occurrence | |||
| Related Issues: |
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| Environment: | Attachments: | ( 0) Patch_11.7_DTLS.zip | |||
| Description: | This is a follow-up ticket that relates to asterisk patched as described here:
Patch 1: ASTERISK-22961 Patch 2: 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 ASTERISK-22961 2) This got us to a working WebRTC on Chrome/Firefox over WS 3) Then, we have applied the patch from ASTERISK-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@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) | ||||
| Comments: | By: Matt Jordan (mjordan) 2014-01-27 18:09:49.355-0600 I'm closing this out as a duplicate of ASTERISK-21930. # Asterisk over WSS doesn't work without Moises's patches. Until that problem is resolved, everything else is pretty much dead in the water: there's no way for someone to accurately work any other issues without patching Asterisk, and even if they fixed them, it still relies on patches from other issues. # Opening a bug that relies on another patch is impossible for someone to adequately triage or work. There is already substantial work and progress being made on ASTERISK-21930; this issue should be addressed there if possible. Once ASTERISK-21930 is closed and committed, this issue can be re-opened if it is still a problem. By: Jay Jideliov (jideliov) 2014-01-27 18:20:48.960-0600 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) By: Jay Jideliov (jideliov) 2014-01-28 17:58:09.824-0600 The issue has been fixed by manually going through all of the patches. I am attaching a combined 11.7 patch that enables WSS calls (DTLS/SRTP). | ||||