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

Fri Sep 13 22:51:59 MSD 2013

That should not matter.  It will be taken care of.

Verify you can just use bandwidth to call the server and run an echo test
or something.
Then try a sip phone registered.  Then on to the webrtc instance.

You probably have some nat problems to bandwidth regardless of WebRTC.

On Fri, Sep 13, 2013 at 1:42 PM, James Mortensen <
james.mortensen at synclio.com> wrote:

> 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
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>
>>>
>>
>
> _________________________________________________________________________
> 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
>
>

-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130913/36788643/attachment.html 