[asterisk-bugs] [JIRA] (ASTERISK-25559) Asterisk12 with webRTC , can ring but no audio and video

[Calvin Leung (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-25559?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Calvin Leung updated ASTERISK-25559:
------------------------------------

    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 at my.public.ip.address
SIP URI: sip:44444 at my.public.ip.address
SIP password: ******
WS URI: ws://my.public.ip.address:8088/ws

  was:
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 at my.private.ip.address
SIP URI: sip:44444 at my.private.ip.address
SIP password: ******
WS URI: ws://my.private.ip.address:8088/ws


> Asterisk12 with webRTC , can ring but no audio and video
> --------------------------------------------------------
>
>                 Key: ASTERISK-25559
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-25559
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>    Affects Versions: 12.8.2
>         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 43
>            Reporter: Calvin Leung
>         Attachments: extensions.conf, http.conf, pjsip.conf, rtp.conf, sip.conf, sip_set_debug_on_20151115_44444-57440_2public.log
>
>   Original Estimate: 15d
>  Remaining Estimate: 15d
>
> 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 at my.public.ip.address
> SIP URI: sip:44444 at my.public.ip.address
> SIP password: ******
> WS URI: ws://my.public.ip.address:8088/ws



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