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Summary:ASTERISK-25559: Asterisk12 with webRTC , can ring but no audio and video
Reporter:Calvin Leung (calvin0029)Labels:
Date Opened:2015-11-15 07:16:00.000-0600Date Closed:2015-11-15 07:18:55.000-0600
Priority:MajorRegression?No
Status:Closed/CompleteComponents:
Versions:12.8.2 Frequency of
Occurrence		Constant
Related
Issues:
Environment:a virtual machine on aliyun with public IP and private IP CentOS6.7 Asterisk 12.8.2 pjproject 2.4.5 libsrtp 1.4.4 jssip 0.7.6 chrome 43Attachments:( 0) extensions.conf
( 1) http.conf
( 2) pjsip.conf
( 3) rtp.conf
( 4) sip_set_debug_on_20151115_44444-57440_2public.log
( 5) sip.conf
Description:Calls between softphones (on cellphone app named Linphone) are OK with both audio and video. But when I call from or to the SIP user on web, I can hear and see nothing from the other side.

I used the demo of jssip on http://tryit.jssip.net/, filled in the form like this:
name: sip:44444@my.public.ip.address
SIP URI: sip:44444@my.public.ip.address
SIP password: ******
WS URI: ws://my.public.ip.address:8088/ws
Comments:By: Asterisk Team (asteriskteam) 2015-11-15 07:16:02.198-0600

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: Joshua C. Colp (jcolp) 2015-11-15 07:18:47.638-0600

Per the Asterisk versions page [1], the maintenance (bug fix) support for the Asterisk branch you are using has ended. For continued maintenance support please move to a supported branch of Asterisk. After testing with a supported branch, if you find this problem has not been resolved, please open a new issue against the latest version of that Asterisk branch.

Thanks!

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions