[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 00:31:53 MSD 2013
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
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>
>>>>
>>>
>>
>> _________________________________________________________________________
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>> http://www.freeswitchsolutions.com
>>
>>
>>
>>
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>>
>
>
> --
> Anthony Minessale II
>
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