[Freeswitch-users] FreeSwitch + WebRTC + JsSIP + Chrome no audio

Sat Oct 5 19:54:42 MSD 2013

Sounds like some part of the audio renderer is not implemented on your version of chrome?

On Oct 4, 2013, at 4:58 PM, Rafael Santana <rafaelstnoliveira at gmail.com> wrote:

> James,
> 
> Yes, Chrome is on a machine at the same network. 
> 
> This log file (http://pastebin.freeswitch.org/21480) is from a test scenario where I tried, using Tryit Demo, call the 5000 extension in FS. On the FS console I can see that the ivr call follow is executed perfectly, but I can't hear anything on my headset. I got this error during this test.
> 
> [11:11:1004/174001:ERROR:webrtc_audio_renderer.cc(241)] Not implemented reached in virtual void content::WebRtcAudioRenderer::Start()
> [11:23:1004/174001:ERROR:platform_thread_linux.cc(99)] Failed to set nice value of thread to -10
> 
> I'm checking for clues right now. Any idea about it? 
> 
> Thanks for the attention!
> 
> 
> 2013/10/4 James Mortensen <james.mortensen at synclio.com>
> But Chrome isn't on the same network, right? Also, I'm not an expert on this, but from what I understand, STUN binding is something that occurs between Chrome and the media server, not a STUN server.  See Example 17 here in this RFC spec: http://tools.ietf.org/html/rfc5245#section-17.  The STUN binding occurs between the two user agents, where one is the SIP user and the other could be your media server.
> 
> Chrome will complain about STUN binding errors or receiving unknown packets. If audio isn't flowing, all I'm trying to say is it might not be a FreeSWITCH issue and you should make sure Chrome isn't the culprit before changing too many things in FreeSWITCH. In one instance, my server's network was the problem, and setting up the same exact FreeSWITCH (and even Asterisk) configuration resulted in two way audio.  :D)
> 
> Not saying this is your problem, just that you should definitely be watching what's happening in the Chrome debug logs too.
> 
> Hope this helps!
> 
> 
> James
> 
> 
> 
> On Fri, Oct 4, 2013 at 12:19 PM, Rafael Santana <rafaelstnoliveira at gmail.com> wrote:
> Thanks for the replies! I couldn't test or check anything today. As soon as I do the tests I will inform here the new status. 
>  
> @James
> My application server (nginx) is on the same network my FreeSwitch and Asterisk are, so I'm not using a Stun server. 
> 
> []'s
> 
> 
> 2013/10/4 James Mortensen <james.mortensen at synclio.com>
> Hi Rafael,
> 
> You didn't mention whether the server was in the cloud.  If you're server is on Amazon EC2, make sure you're following the guide here:  https://wiki.freeswitch.org/wiki/Amazon_EC2
> 
> Also, if you run a tcpdump -s0 -v udp on your FreeSWITCH and Asterisk server, do you see audio flowing?  Also, in Chrome, startup chrome from the command line with the options to enable debug logging:
> 
> chrome --enable-logging --v=11
> 
> Then look to see if there are STUN binding errors.  Also, check chrome://webrtc-internals, which will also tell you if Chrome is trying to send audio.
> 
> Is the server behind NAT or is it on the public Internet with it's own public IP bound to the eth0 interface?  
> 
> Hope this helps!
> 
> 
> 
> James
> 
> 
> 
> On Thu, Oct 3, 2013 at 4:44 PM, Rafael Santana <rafaelstnoliveira at gmail.com> wrote:
> Hi,
> 
> I'm new to telephony and FreeSwitch's world, so I apologize in advance for any nonsense I speak here. 
> 
> I've been trying to setup an environment where It can be possible to make a call through Google Chrome Browser using JsSIP to a standard phone device on PSTN. 
> 
> In my network my "PSTN gateway" is an Asterisk 1.4 instance (No, I can't chance it today). To communicate with Chrome I have a FreeSwitch 1.5.5 instance and to get access to PSTN via this instance I had to register my Asterisk instance as a gateway on my Sofia's external profile. This part of my scenario works fine. I'm able to make calls using a softphone registered on FreeSwitch to standard phones on PSTN with no problems. What I wasn't able to do until now was the JsSIP + FreeSwitch integration.      
> 
> To setup FreeSwitch to comunicate with JsSIP, the only thing I did was uncomment the line below on sip_profiles/internal.xml.
> 
> <param name="ws-binding" value=":5066"/>
> 
> I really don't know if just this is sufficient. Am I missing something important?
> 
> To connect on my FreeSwitch instance from Chrome, I'm using the Tryit JsSIP demo. Today, I'm able to register on FS from Tryit demo and perform a call to a PSTN phone. The connection is established but I don't get any audio in both endpoints. The same happens when I try to call the 5000 ivr extension or an user on a softphone at the same network from my Chrome browser.
> 
> Assuming that all the services I've mentioned here are running on the same network, do you have any idea why I can't get audio in both endpoints of my experiment?
> 
> Additional information:
> Ubuntu 12.04 64 bits
> FreeSwitch version 1.5.5 default install configuration
> Tryit JsSIP Demo with jssip-0.3.0.js
> 
> Thanks in advance,
> Rafael.
> 
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> 
> -- 
> Rafael Santana Oliveira
> Mestre em Ciência da Computação
> 
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> 
> 
> -- 
> Rafael Santana Oliveira
> Mestre em Ciência da Computação
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
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