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

Tue Oct 8 22:26:02 MSD 2013

Hi guys! Thanks for your help! I got it running right now.

For future reference, I did the following steps to accomplish a two way
audio while calling from Chrome to a PSTN phone.

I've migrated my environment to a CentOS 5.9. I've installed FreeSwitch
following the default recipe, but instead of

./configure -C

 I did

./configure --with-openssl # suggested by @Iwan

To make available websocket connection on FS, I've just uncommented the
following on sip_profiles/internal.xml

<!-- uncomment for sip over websocket support -->
> <!--<param name="ws-binding"  value=":5066"/> -->

And that is it! :D

To reach PSTN I used a pre-configured Asterisk instance as gateway.

It is important to note that my client (Chrome), FreeSwitch and Asterisk
were all at the same network while doing this tests.

PS: I turned off my iptables to eliminate any possibility of port blocking
while doing my tests.

Thanks again!

2013/10/6 Steven Ayre <steveayre at gmail.com>

> STUN allows a client behind NAT to find the IP:port its packets are
> leaving externally on so that it knows the location to tell the server
> (FreeSWITCH) to send audio back to.
>
> In short STUN is used at whichever end is using NAT (which could be none,
> one or both).
>
> If FreeSWITCH is on a public IP but your PC running Chrome is on NAT
> (extremely likely) then Chrome still needs to use STUN.
>
>
> On 4 October 2013 21:21, James Mortensen <james.mortensen at synclio.com>wrote:
>
>> 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
>>>>>
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>>>>> 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
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>>>>
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>>>>
>>>>
>>>
>>>
>>> --
>>> Rafael Santana Oliveira
>>> Mestre em Ciência da Computação
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
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>>>
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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
>>
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>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
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>
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>

-- 
Rafael Santana Oliveira
Mestre em Ciência da Computação
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