Webrtc for Vicidial

[iboam]

Im able to make calls using https://www.doubango.org/sipml5/call.htm but now it's not even ringing on vicidial

[iboam]

Im able to make calls using https://www.doubango.org/sipml5/call.htm and https://tryit.jssip.net/ but on vicidial it's not even ringing when login as agent

[williamconley]

Im able to make calls using https://www.doubango.org/sipml5/call.htm and https://tryit.jssip.net/ but on vicidial it's not even ringing when login as agent

Since you did not post any asterisk output, log errors, configuration changes ...

I would say that you have found out why there are presently two different groups creating WebPhones that DO work for Vicidial. Cuz it's not "just plug it in and it'll work". Wait for one of them to finish and/or drop some $$ on them to get involved in the development or help to speed it along.

[iboam]

I'm just following his instructions @ https://github.com/chornyitaras/PBXWebPhone/wiki

[williamconley]

In that case, based on the fact that someone else did it and got a ring but no answer ... and we've never looked at it ... I'd say wait for The Vicidial Group's version to come out. It'll likely be an upgrade to Vicibox, but it may require a full re-install.

[chornyi_taras]

Im able to make calls using https://www.doubango.org/sipml5/call.htm and https://tryit.jssip.net/ but on vicidial it's not even ringing when login as agent

Please update PBXWebPhone to latest version and clear your browser cache

Do you see webphone frame when login as agent?

should be something like this

If not please provide browser logs(press F12 and select console). Also please make sure that you are using https.
Output of asterisk command "sip show peer <phone>" will be helpful.

[iboam]

Trust me i did and reviewed every single line ... i'm getting this error in vicidial An Error occurred while connecting to the websocket.

[williamconley]

If not please provide browser logs(press F12 and select console). Also please make sure that you are using https.
Output of asterisk command "sip show peer <phone>" will be helpful.

[chornyi_taras]

Trust me i did and reviewed every single line ... i'm getting this error in vicidial An Error occurred while connecting to the websocket.

Specify Web Socket URL:

Admin->Servers-><Dialler server>-> Web Socket URL: to "wss://<server domain name>:8089/ws"

[mcargile]

If you are interested in testing our WebRTC phone for Vicidial you should check out this thread:

viewtopic.php?f=3&t=35875

[arpit.modi]

We have followed steps from https://github.com/chornyitaras/PBXWebPhone and configured this module and it seems to be working fine except audio issue.

My system information is as below:

Admin Vicidial Information
VERSION: 2.14-611a
BUILD: 170425-1353
© 2017 ViciDial Group

Agent Vicidial Information
VERSION: 2.14-520c
BUILD: 170416-1640

Asterisk Version
Asterisk 11.25.1-vici

Log for rtp is as below:

[Apr 28 03:33:33] Sent RTP packet to xxx.xxx.xxx.xxx:58125 (type 00, seq 033387, ts 480640, len 000160)
[Apr 28 03:33:33] Sent RTP packet to xxx.xxx.xxx.xxx:58125 (type 00, seq 033388, ts 480800, len 000160)
[Apr 28 03:33:33] Sent RTP packet to xxx.xxx.xxx.xxx:58125 (type 00, seq 033389, ts 480960, len 000160)
[Apr 28 03:33:33] Sent RTP packet to xxx.xxx.xxx.xxx:58125 (type 00, seq 033390, ts 481120, len 000160)
[Apr 28 03:33:33] Sent RTP packet to xxx.xxx.xxx.xxx:58125 (type 00, seq 033391, ts 481280, len 000160)
[Apr 28 03:33:33] Sent RTP packet to xxx.xxx.xxx.xxx:58125 (type 00, seq 033392, ts 481440, len 000160)
[Apr 28 03:33:33] Sent RTP packet to xxx.xxx.xxx.xxx:58125 (type 00, seq 033393, ts 481600, len 000160)

When agent is logging in, and press call Agent webphone:

[Apr 28 06:57:14] == Using SIP RTP CoS mark 5
[Apr 28 06:57:15] > Channel SIP/5555-00000002 was answered
[Apr 28 06:57:15] -- Executing [8600052@default:1] MeetMe("SIP/5555-00000002", "8600052,F") in new stack
[Apr 28 06:57:15] == Parsing '/etc/asterisk/meetme.conf': Found
[Apr 28 06:57:15] == Parsing '/etc/asterisk/meetme-vicidial.conf': Found
[Apr 28 06:57:15] -- Created MeetMe conference 1023 for conference '8600052'
[Apr 28 06:57:15] -- <SIP/5555-00000002> Playing 'conf-onlyperson.gsm' (language 'en')

[Apr 28 06:57:46] NOTICE[7292]: chan_sip.c:29370 check_rtp_timeout: Disconnecting call 'SIP/5555-00000002' for lack of RTP activity in 61 seconds
[Apr 28 06:57:46] -- Hungup 'DAHDI/pseudo-1305219282'

[Apr 28 06:57:46] == Spawn extension (default, 8600052, 1) exited non-zero on 'SIP/5555-00000002'
[Apr 28 06:57:46] -- Executing [h@default:1] AGI("SIP/5555-00000002", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----44---------------") in new stack

It looks like issue with RTP, we have already opened rtp port range from firewall.

Can you please suggest me how to make it working?
Is there anything wrong in configuration?

Just to add up, It works fine with any other softphone and audio coming properly. So it must be something from WebRTC side i guess.

Any help would be highly appreciated.

[chornyi_taras]

add stun server config to
/etc/asterisk/rtp.conf

Code: Select all

icesupport=yes
stunaddr=stun.l.google.com:19302


[arpit.modi]

Hi,

Thanks for your response. I have already have it in my rtp.conf
But still no audio.

Any other ideas?

[williamconley]

Temporarily (for just the test) turn off the firewall completely.

also: Does the server have a public IP? If not, this can present a challenge for SIP to SIP. If your client and your server are both behind a firewall, challenges can ensue.

Verify the externip value in sip.conf of course.

[arpit.modi]

Thanks for your response.
Yes, externip and other settings are perfect.

Actually the issue was something strange. It worked fine when the server was on public IP but when we put behind firewall and do port forward it was having RTP issue.

Somehow it got fixed by changing NAT policy to manual in firewall settings.
Still checking more in different scenarios.

Thanks for support.

[williamconley]

That's not "strange", that's what happens to SIP when you don't maintain a clear NonNAT pathway. Drama ensues.

[arpit.modi]

Yeah I completely understand. Firewall and networking is absolutely not my cup of tea.
Thanks for helping out.