Skip to content

One way video, any browser, webrtc2sip and Asterisk to softphone #228

@GoogleCodeExporter

Description

@GoogleCodeExporter
The problem occurs when I make or receive a call from Firefox or Chrome, via 
webrtc2sip and asterisk to a softphone (Ekiga). I've attached the sipml5 log 
file.

I think I'm getting audio in both directions (the mic is faulty on one 
machine), but video only in one direction. Video is sent from the web browser 
to the softphone, but doesn't get back to the web browser. There are messages 
appearing in the webrtc2sip debug:


*[DOUBANGO INFO]: Incoming SIP INFO(picture_fast_update)
*[DOUBANGO INFO]: ***IDR request tooo close(2 ms)...ignoring****


and 

**[DOUBANGO WARN]: function: "tdav_codec_h263_decode()"
file: "src/codecs/h263/tdav_codec_h263.c"
line: "440"
MSG: Buffer overflow


Also, before video starts I get thousands of 

*[DOUBANGO INFO]: Sending DNS query to "192.168.1.250"
*[DOUBANGO INFO]: Sending DNS query to "192.168.1.250"
*[DOUBANGO INFO]: Sending DNS query to "192.168.1.250"
*[DOUBANGO INFO]: Sending DNS query to "192.168.1.250"


appearing in the webrtc2sip log.

The asterisk config allows g722 and h263 only.

It's probably something that isn't configured correctly, or a mistake in my 
sipml5 code, but I'd be grateful for some help solving this problem.

thanks,

Charles.












What steps will reproduce the problem?
1.
2.
3.

What is the expected output? What do you see instead?


What version of the product are you using? On what operating system?


Please provide any additional information below.


Original issue reported on code.google.com by charles....@gmail.com on 15 Jun 2015 at 3:53

Attachments:

Metadata

Metadata

Assignees

No one assigned

    Projects

    No projects

    Milestone

    No milestone

    Relationships

    None yet

    Development

    No branches or pull requests

    Issue actions