| Summary: | ASTERISK-27636: One way audio on Chrome 64 webrtc chan_sip | ||
| Reporter: | Sebastian Gutierrez (sum) | Labels: | webrtc |
| Date Opened: | 2018-01-29 09:57:53.000-0600 | Date Closed: | 2018-02-06 09:35:21.000-0600 |
| Priority: | Major | Regression? | |
| Status: | Closed/Complete | Components: | Channels/chan_sip/General |
| Versions: | 13.19.0 | Frequency of Occurrence | Constant |
| Related Issues: | |||
| Environment: | Ubuntu 16.04 | Attachments: | ( 0) NOTOK.txt ( 1) OK.txt ( 2) sipdebug.txt |
| Description: | After updating chrome to 64 (didn't happen on 63) we are having one way audio for an inbound call, does not happen on outbound calls.
WebRTC, JSSIP I think may be related to some issue on the sdp, I will attach now the sip debug client side that is possible to see one with a chrome 63 that works and the other with 64 that wont work ok, I will upload a server side log | ||
| Comments: | By: Asterisk Team (asteriskteam) 2018-01-29 09:57:54.495-0600 Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution. A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report. Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process]. By: Asterisk Team (asteriskteam) 2018-01-29 09:57:55.464-0600 The module you are reporting the issue against is no longer supported as a core module but your issue is in the queue. Your patience is appreciated as a community developer may work the issue when time and resources become available. Asterisk is an open source project and community members work the issues on a voluntary basis. You are welcome to develop your own patches and submit them to the project.[1] If you are not a programmer and you are in a hurry to see a patch provided then you might try rallying support on the Asterisk users mailing list or forums.[2] Another alternative is offering a bug bounty on the asterisk-dev mailing list.[3] Often a little incentive can go a long way. [1]: https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process [2]: http://www.asterisk.org/community/discuss [3]: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Bug+Bounties By: Sebastian Gutierrez (sum) 2018-01-29 09:59:20.813-0600 JSSIP with sip log By: Sebastian Gutierrez (sum) 2018-01-29 10:52:27.716-0600 asterisk side By: Sebastian Gutierrez (sum) 2018-02-06 09:35:21.249-0600 I´ve changed the way the stream is added and that fixed the problem, so I think the sdp is correct.... | ||