What steps will reproduce the problem?
1. Architecture: Chrome/sipml(Client side) <-> Asterisk <-> GenesysSIP <->
webrtc2sip <-> Chrome/sipml (Agent side)
2. When agents log with webbreaker=false, everythink is fine, but we need also
calls from PSTN network and we are enforced to log agents with webbreaker=true
3.When webbreaker=true, calls from client side (browser/Chrome/sipml) are not
established correctly.
What is the expected output? What do you see instead?
I expect calls will be correctly established when webbreaker is enabled.
However there is no audio and video. SIP session is correct I think.
What version of the product are you using? On what operating system?
webrtc2sip - 2.6.0
sipml5 - 1.3.203
Chrome - 34
Please provide server logs with DEBUG level equal to INFO
Attached files:
'disabled' - call debug when webbreaker=false
'enabled' - call debug when webbreaker=true
Please provide browser logs
I think no need because I see that calls have two different flows on webrtc2sip
side when webbreaker is enabled/disabled. I think there is a problem.
In logs I see two problems:
***ERROR: function: "tsk_params_get_param_value()"
file: "src/tsk_params.c"
line: "219"
MSG: Invalid parameter
I dont have any idea what does it mean.
And:
**WARN: function: "tdav_session_av_prepare()"
file: "src/tdav_session_av.c"
line: "422"
MSG: DTLS-SRTP requested but no SSL certificates provided, disabling this
option :(
I have SRTP on both side so I dont understand why certifiacate is needed in
this place, but this is only warning.
Thanks for any help!
Original issue reported on code.google.com by gdziarm...@gmail.com on 5 Jun 2014 at 7:57
Original issue reported on code.google.com by
gdziarm...@gmail.comon 5 Jun 2014 at 7:57Attachments: