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Outbound call hangs up after 1 minute with FreeSwitch Session-Expire #224

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) cannot find duplicate, but this is similar issue: 
http://lists.freeswitch.org/pipermail/freeswitch-dev/2009-February/002012.html
c) console attached

What steps will reproduce the problem?
1. Make server for WebRTC with Freeswitch
a) compile freeswitch 
https://freeswitch.org/confluence/display/FREESWITCH/CentOS+6
b) setup for webrtc: https://freeswitch.org/confluence/display/FREESWITCH/WebRTC
2. Setup freeswitch for outbound dialing obviously
3. Dial out from sipml5 to a cell phone (version svn 222 and 230 both have same 
issue)
4) after one minute, freeswitch will send an INVITE as a keep alive, and this 
causes the following error: (it seems like SIPML5 is trying to adjust 
properties on an existing call) "Failed to set local answer sdp: Session error 
code: ERROR_CONTENT. Session error description: Failed to set audio receive 
codecs.."

What is the expected output? What do you see instead?
Expected is it shouldn't err out, and the call should continue.
I see immediately after that error it sends a BYE hanging up the call. 
Error at line 444 of attachment.

What version of the product are you using? On what operating system?
SVN 222 + 230 [tried both], windows 7, chrome 41 
also tried svn 230 (sipML5-v1.2015.03.18?) on chrome 42.0.2311.135 with same 
result.

Please provide any additional information below.
I tried this with svn 222 and it failed, so i went to 230 and same problem.
I would like to just disable the line attempting to change the codec on the 
channel bypassing it and hopefully it will continue working, but I haven't been 
able to track it down yet. 

I replaced my personal public ip with this: {mylocalip} 
everything else in there should be self-explanatory

It looks like SIPML5 isn't setting a session-expires header in the INVITE 
itself, but freeswitch is sending one after 60 seconds (half of 120 second 
default), and when that happens, it causes SIPML5 to crash, dropping the call.]

Inbound calls work great, I've sustained calls of over 5 minutes, and this 
problem doesn't happen.

I also tried calling in while I was on the phone, the call didn't crash, but 
the screen turned grey, and I was unable to do anything until i refreshed the 
page, but the call was ok.

Original issue reported on code.google.com by the.at.r...@gmail.com on 9 May 2015 at 2:33

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