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

Sat Oct 5 00:58:46 MSD 2013

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.
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
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>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
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>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Rafael Santana Oliveira
>> Mestre em Ciência da Computação
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

-- 
Rafael Santana Oliveira
Mestre em Ciência da Computação
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