fangjz / sipml5

Automatically exported from code.google.com/p/sipml5
BSD 3-Clause "New" or "Revised" License
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Static on one end of the call #116

Open GoogleCodeExporter opened 9 years ago

GoogleCodeExporter commented 9 years ago
a) Before posting your issue you MUST answer to the questions otherwise it
will be rejected (invalid status) by us
b) Please check the issue tacker to avoid duplication
c) Please provide network capture (Wireshark) or Javascript console log
if you want quick response

What steps will reproduce the problem?
1. Following the steps on the wiki page
2. Using asterisk for all the addressing? 
3. Calling anywhere using the asterisk trunk. Chrome has audio from the called 
party, but the called party has only static. Also happens in chrome->chrome 
calls. 

What is the expected output? What do you see instead?
2 way audio. Instead I get 1 way audio and static on the otherside. Also 
happens when calling asterisk (voicemail extension, etc). 

What version of the product are you using? On what operating system?
The demo on your website, it says 179. Happening on Linux and windows using the 
latest version of chrome. 

Please provide any additional information below.

New to WebRTC, spent the last week or so learning about it and getting it setup 
and reading tutorials, etc. I have asterisk working fine and it works from 
endpoint to endpoint and even from client to phone.. 

Original issue reported on code.google.com by Jimster...@gmail.com on 24 Jul 2013 at 10:10

GoogleCodeExporter commented 9 years ago
Fixed the problem. The trunked asterisk version in the tutorial doesnt work 
with the newest chrome. 

Original comment by Jimster...@gmail.com on 2 Aug 2013 at 7:16