[Freeswitch-users] Configuring Freeswitch 1.4b for WebRTC Peer to Peer and to the PSTN
James Mortensen
james.mortensen at synclio.com
Sat Sep 14 02:13:18 MSD 2013
Hi Anthony,
All the ports were open, but I didn't have the external IP and the settings
listed in the Amazon EC2 documentation set. After setting them and
restarting Freeswitch, I made 20 successful calls in a row with 2 way audio
from Chrome to the PSTN!
Thank you for your help!
James
James
*"Every Call, Every Time!"*
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On Fri, Sep 13, 2013 at 1:42 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:
> did you do all the things in https://wiki.freeswitch.org/wiki/Amazon_EC2
>
> Also are you using the external or internal profile (port 5065 is the
> default unauthenticated port)
> if you want to use internal you have to comment out the auth-calls and the
> inbound acl lines.
>
> Most of this should be available in our various books and wiki articles.
>
>
>
> On Fri, Sep 13, 2013 at 3:31 PM, James Mortensen <
> james.mortensen at synclio.com> wrote:
>
>> Hi Anothony,
>>
>> Inbound calling is proving to be more challenging. Here's what I've
>> tried from https://wiki.freeswitch.org/wiki/Auto_NAT:
>>
>> 1. Verified the following params were set in my sip profiles (these were
>> already set by default):
>>
>>
>>
>> <param name="ext-rtp-ip" value="auto-nat"/>
>> <param name="ext-sip-ip" value="auto-nat"/>
>>
>>
>> 2. Ran sofia status, which shows this:
>>
>>
>> Name Type
>> Data State
>>
>> =================================================================================================
>> 10.166.245.111 alias
>> internal ALIASED
>> internal-ipv6 profile sip:mod_sofia@
>> [::1]:5060 RUNNING (0)
>> external profile
>> sip:mod_sofia at 10.166.245.111:5080 RUNNING (0)
>> external::example.com gateway
>> sip:joeuser at example.com NOREG
>> internal profile
>> sip:mod_sofia at 10.166.245.111:5060 RUNNING (0)
>> internal::bandwidth.com gateway sip:your user
>> name at 67.231.8.195 NOREG
>>
>> =================================================================================================
>> 3 profiles 1 alias
>>
>>
>> 3. Ran *sofia status profile internal* expecting to see ext-rtp-ip and
>> ext-sip-ip set, but they're missing:
>>
>> Name internal
>> Domain Name N/A
>> Auto-NAT false
>> DBName sofia_reg_internal
>> Pres Hosts 10.166.245.111,10.166.245.111
>> Dialplan XML
>> Context public
>> Challenge Realm auto_from
>> RTP-IP 10.166.245.111
>> SIP-IP 10.166.245.111
>> URL sip:mod_sofia at 10.166.245.111:5060
>> BIND-URL sip:mod_sofia at 10.166.245.111:5060;transport=udp,tcp
>> HOLD-MUSIC local_stream://moh
>> OUTBOUND-PROXY N/A
>> CODECS IN G722,PCMU,PCMA,GSM
>> CODECS OUT G722,PCMU,PCMA,GSM
>> TEL-EVENT 101
>> DTMF-MODE rfc2833
>> CNG 13
>> SESSION-TO 0
>> MAX-DIALOG 0
>> NOMEDIA false
>> LATE-NEG true
>> PROXY-MEDIA false
>> ZRTP-PASSTHRU true
>> AGGRESSIVENAT false
>> CALLS-IN 86
>> FAILED-CALLS-IN 86
>> CALLS-OUT 0
>> FAILED-CALLS-OUT 0
>> REGISTRATIONS 1
>>
>>
>> So regarding NAT issues, it seems Freeswitch isn't able to get the
>> server's public IP perhaps, despite auto-nat being the correct setting?
>> This is just an EC2 server behind Amazon NAT.
>>
>> Also, I ran:
>> nat_map status
>>
>> Nat Type: UNKNOWN, ExtIP:
>> NAT port mapping enabled.
>>
>> 0 total.
>>
>>
>> I did perform an echo test from user 1000 to 9196 from Chrome to
>> Freeswitch, which works as expected, but getting Bandwidth incoming calls
>> to be processed correctly is a challenge. It just loops over and over again
>> with this in the DEBUG console:
>>
>> 2013-09-13 20:27:31.255055 [NOTICE] switch_channel.c:1030 New Channel
>> sofia/internal/+19712383780 at 192.168.37.68[80196f26-febf-4363-b48c-ee88022c6b9c]
>> 2013-09-13 20:27:31.255055 [DEBUG] switch_core_session.c:1006 Send signal
>> sofia/internal/+19712383780 at 192.168.37.68 [BREAK]
>> 2013-09-13 20:27:31.255055 [DEBUG] switch_core_session.c:1006 Send signal
>> sofia/internal/+19712383780 at 192.168.37.68 [BREAK]
>> 2013-09-13 20:27:31.255055 [DEBUG] switch_core_state_machine.c:418
>> (sofia/internal/+19712383780 at 192.168.37.68) Running State Change CS_NEW
>> 2013-09-13 20:27:31.255055 [DEBUG] switch_core_state_machine.c:436
>> (sofia/internal/+19712383780 at 192.168.37.68) State NEW
>> 2013-09-13 20:27:31.275057 [DEBUG] sofia.c:8003 IP 67.231.4.195 Rejected
>> by acl "domains". Falling back to Digest auth.
>> 2013-09-13 20:27:31.275057 [DEBUG] switch_core_session.c:1006 Send signal
>> sofia/internal/+19712383780 at 192.168.37.68 [BREAK]
>> 2013-09-13 20:27:31.275057 [DEBUG] sofia.c:1787 detaching session
>> 80196f26-febf-4363-b48c-ee88022c6b9c
>>
>>
>> So in summary, the WebRTC seems good but not the PSTN. I can do Chrome to
>> FS to Chrome, Chrome to FS Echo, I can call from Chrome to FS to Bandwidth
>> (NO AUDIO) but calling inbound loops. I'm assuming all of the routing is
>> done using the Bandwidth DID I've pointed to the server, not the 971xxxxxxx
>> number that I'm calling from on Verizon/Google Voice.
>>
>> Thank you for your help!
>> James
>>
>>
>> On Fri, Sep 13, 2013 at 11:51 AM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> 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
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>>> googletalk:conf+888 at conference.freeswitch.org
>>> pstn:+19193869900
>>>
>>> _________________________________________________________________________
>>> 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
>>
>>
>>
>>
>> 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
>
> _________________________________________________________________________
> 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
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