UPD flows bidirectionally but one way audio on webphone

[GoogleCodeExporter]

What steps will reproduce the problem?
1. Log in the webphone SIPML5
2. Launch a call from webphone
3. Answer the call

What is the expected output? What do you see instead?
It should have two-way audio, but in the webphone I dont have audio

What version of the product are you using? On what operating system?
Asterisk 11.9
webRTC2SIp 
SIPML5 1.4.217
Chrome 35.0.1916.153 m

Windows7 & Windows8

Please provide any additional information below.

When I launch calls, sometimes I have one way audio, but the thing is that the 
wireshark shows UDP flowing in both sides. For some reason, chrome does not 
"take" this packets.

I provide logs from everything.

Thanks in advance.

Original issue reported on code.google.com by alejandr...@gmail.com on 24 Jun 2014 at 3:57

Attachments:

• capWebphone.pcapng
• [WebRTC Internals.pdf](https://storage.googleapis.com/google-code-attachments/sipml5/issue-185/comment-0/WebRTC Internals.pdf)

[GoogleCodeExporter]

We have the same issue, slightly different setup.

We're using FreeSWITCH v1.4.x . WebRTC works fine most of the time, but 
sometimes we see audio flowing to Chrome (and Chrome's own WebRTC internals 
indicate the stream is up) but no audio is heard. Microphone works and the 
remote party can hear. This is not a NAT issue from the looks of it, as the UDP 
is flowing and the ports look right.

Original comment by RdFlt...@gmail.com on 29 Jul 2014 at 6:09

[GoogleCodeExporter]

For me all problems were solved putting Kamailio + mediaproxy-ng in front of 
asterisk.

Original comment by virmanta...@lamoda.ru on 30 Jul 2014 at 6:23

[GoogleCodeExporter]

Interestingly, we tested the exact same environment using sip.js and can't 
reproduce the issue there. I know that's not super helpful but we were getting 
desperate :-)  This sort-of narrows it down to something in the JS so would 
love some guidance here if possible.

Original comment by RdFlt...@gmail.com on 30 Jul 2014 at 9:18