[Freeswitch-users] Configuring Freeswitch 1.4b for WebRTC Peer to Peer and to the PSTN

[James Mortensen]

Here is my follow up to this issue.  It seems something happened to the
server where it stopped sending Binding responses to Chrome.  This is a
known issue that I've seen before that Google says was an Asterisk issue.
 However, the same thing happens to Freeswitch as well, indicating the
problem is the network/server, not the software.

I booted up another server from a snapshot, verified two way audio with
Asterisk, then reinstalled Freeswitch, uncommented the ws-binding parameter
to enable WS on port 5066, then registered user 1000 and 1002 from the
tryit.jssip.net demo using the following configuration:

Name: James
SIP URI:  sip:1000 at 54.X.X.X   <-- public IP of server
password: 1234
WS URI: ws://54.X.X.X:5066  <--- same public IP

and I substituted 1002 in place of 1000 for another user on another
network.  I verified audio flows both ways with audio flowing through
Freeswitch.

I made no other changes to the configuration.  This was a lot easier to get
started with than Asterisk WebRTC.  I just made it out to be a lot harder
than it was by assuming I needed to manually add in my server's IP address
in the configuration files.  This happens automatically, even on a NAT'd
server.  Amazing! :)


Now, for the PSTN part, I configured my BANDWIDTH.com provider as an
internal context and as an outbound and inbound dialplan.  I can connect a
call between my cellphone and Chrome, but there's no audio flowing to/from
Bandwidth.  I suspect the problem has something to do with bridging AVP and
SAVPF, since the carriers don't support SRTP.

I've grepped the configuration files, and I don't see how I would configure
the system to bridge AVP and SAVPF and do transcoding.  Any ideas?

Thank you!!

James



On Wed, Sep 11, 2013 at 12:01 PM, James Mortensen <
james.mortensen at synclio.com> wrote:

> Installing ibncursesw5 and libncursesw5-dev did resolve the issue.  I can
> now connect to the WebSocket server.  Seems the cluechoo module is just
> something that was added in to separate the help vampires from the people
> with legitimate issues. :D
>
> I got 405 Method Not Allowed errors, and I resolved them by making sure
> the IP address in the SIP URI matches the IP address in the WS field.
>
> At this time, I've successfully registered and am getting back 200 OK's
> and have now moved onto tweaking the settings so I can get audio going both
> ways. The candidates appear to be only showing the public IP, so I'll have
> to figure that out. I'll provide more updates, or questions, as I move
> forward.
>
> James
>
>
> On Wed, Sep 11, 2013 at 11:32 AM, James Mortensen <
> james.mortensen at synclio.com> wrote:
>
>> Okay, I added in the dependencies and still get those errors.  Do I need
>> to file a bug in JIRA for this or am I just missing something?
>>
>> I'm happy to try a different OS if there's one that's been tried and
>> tested.
>>
>> Of course, more googling reveals that mod_cluechoo is just a joke?
>> http://wiki.freeswitch.org/wiki/Mod_cluechoo
>>
>> > SL (Steam Locomotive) runs across your terminal when you type "sl" as
>> you meant to type "ls". It's just a joke command, and not usefull at all.
>> Put the binary to /usr/local/bin.
>>
>> I hope I'm not chasing a problem that has absolutely no impact on my
>> ability to get WebRTC working here. :D
>>
>>
>> Thanks for any additional help you can provide,
>> James
>>
>>
>>
>> On Wed, Sep 11, 2013 at 11:19 AM, James Mortensen <
>> james.mortensen at synclio.com> wrote:
>>
>>> I ran the netstat command, and it doesn't appear to be listening. It
>>> doesn't appear to be listening on anything.  I do have this error in the
>>> console when starting:
>>>
>>> 2013-09-11 18:16:14.445921 [ERR] switch_nat.c:201 Error checking for PMP
>>> [general error]
>>>
>>> AND
>>>
>>> 2013-09-11 18:09:30.233568 [CRIT] switch_loadable_module.c:1383 Error
>>> Loading module /opt/freeswitch-1.4b/mod/mod_cluechoo.so
>>> **/opt/freeswitch-1.4b/mod/mod_cluechoo.so: undefined symbol: waddch**
>>>
>>> I  believe I overlooked these earlier. But it's possible I'm missing a
>>> dependency.  I'm going to install the libncurses packages as described here
>>> http://jira.freeswitch.org/browse/FS-3689 and then rebuild to see if
>>> that helps.  I'm running Ubuntu 12.10.
>>>
>>> James
>>>
>>>
>>>
>>> On Wed, Sep 11, 2013 at 11:07 AM, Anthony Minessale <
>>> anthony.minessale at gmail.com> wrote:
>>>
>>>> Did you open all the necessary firewall ports?
>>>> Playing around on amazon as your first try complicates thing a bit for
>>>> you.
>>>>
>>>> You should be able to verify its listening on the port with netstat -an
>>>> | grep 5066
>>>>
>>>>
>>>>
>>>> On Wed, Sep 11, 2013 at 12:55 PM, James Mortensen <
>>>> james.mortensen at synclio.com> wrote:
>>>>
>>>>> Here's another update to my adventures in Freeswitch WebRTC.  I assume
>>>>> from the getting started documentation that there are users created by
>>>>> default with the password 1234, so I'm trying to create the ws 5066
>>>>> connection from the TryIt JsSIP demo:  http://tryit.jssip.net
>>>>>
>>>>> Name: James
>>>>> SIP URI: 1000 at Y.Y.Y.Y   <--- Local IP of EC2 server
>>>>> SIP password: 1234
>>>>> WS URI:  ws://X.X.X.X:5066   <--- Public IP of EC2 server
>>>>>
>>>>> Hope this helps!
>>>>> James
>>>>>
>>>>>
>>>>>
>>>>>
>>>
>>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130913/7d4e25d0/attachment.html