WEBRTC Sip server - PBX

h.fabien · Posted: Tue Dec 22, 2020 8:09 am Post subject: WEBRTC Sip server - PBX

1. Brekeke Product Name and Version:
Brekeke PBX 3.10.4.3/517-11

2. Java version:
1.8.0_271

3. OS type and the version:
Windows Server 2016 (10)

4. UA (phone), gateway or other hardware/software involved:
Webrtc : IM-client/OMA1.0 sipML5-v1.2016.03.04
Webrtc : JsSIP 3.5.5
Webrtc using WSS
PhonerLite 2.84

5. Your problem:
Hi,
Cannot receive or make call from webrtc client to sip softphone or voice gateway.
Error : SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/WSS

Webrtc to Webrtc : OK
Webrtc to sip software : KO
504 Time Out - JsSIP 3.5.5 - PhonerLite 2.84 - Inviting
Sip software to Webrtc : KO
488 Failure - PhonerLite 2.84 - JsSIP 3.5.5 - Closing
Webrtc to Voice Gateway : KO
488 Failure - IM-client/OMA1.0 sipML5-v1.2016.03.04 - Cisco-SIPGateway/IOS-16.6.6 - Provisional
Sip software to Voice Gateway : OK

RTP Relay is ON

Any ideas or recommandations ?

Thanks
Fabien

Harold · Posted: Tue Dec 22, 2020 8:39 pm Post subject:

Have you set up the PBX to use WebRTC?
Refer to the following wiki topic.
https://docs.brekeke.com/pbx/setting-up-brekeke-pbx-to-user-webrtc

h.fabien · Posted: Mon Dec 28, 2020 2:17 am Post subject:

Hi,

Thank you, I follow the wiki topic.
I don't understand relationship between User PBX and User authentication.
Anyway, the step to Setting up Brekeke PBX to user WebRTC was followed.
Error seems with codec negotiated, perhaps it's the wrong way.

Fabien

o.mahmoud · Posted: Fri Feb 05, 2021 12:52 pm Post subject:

Any update for this please ?
I have the same issue.

Harold · Posted: Sun Feb 07, 2021 5:48 pm Post subject:

Are you making a call from non-webrtc sip client to webrtc clinet or vice versa?
If so you need to pass a call via Brekeke PBX by DialPlan to convert codecs and SDP.

Tata · Brekeke Master Guru Joined: 27 Jan 2008 Posts: 223

Hi h.fabien and o.mahmoud,
Did you add new DialPlan rules or modify default rules?
You need to keep the default "From PBX" and "To PBX" rules applied to bridge SIP calls for WebRTC client such as JsSIP.

If you added any new DialPlan rules, let you disable them to apply default rules.

o.mahmoud · Posted: Wed Feb 17, 2021 10:27 am Post subject:

I resolved my issue :

I adjusted my dial plan and I kept the target params under ARS empty.

Also, I created a user extension and attached it to a webrtc phone type.