WebRTC - 603 Decline

1. Brekeke Product Name and Version: Brekeke PBX, Version 3.10.5.4

2. Java version: 11.0.10

3. OS type and the version: Windows Server 2019, 10.0

4. UA (phone), gateway or other hardware/software involved:
sipjs
asterisk

5. Your problem: I'm attempting to go from WebRTC (sipjs) to Non-WebRTC (asterisk). In doing so I'm getting a 603 Decline. Looking at the session log it appears like it might be trying to do WebRTC to WebRTC connection because it's adding port 443 to the UAS Address? Everything is currently on same private networks so shouldn't be causing NAT issues.

6 sip:200@10.50.100.73 sip:D121@10.250.95.121 00:00:00.000 2021-03-24 19:15:55.731 2021-03-24 19:15:55.732 WSS:connect timed out 603 10.50.100.64:53636 10.250.95.121:443 Error D121 & WSS-failed SIP.js/0.17.1 Closing

Wed Mar 24 2021 19:15:54 GMT-0400 (Eastern Daylight Time) | sip.Transport | Received WebSocket text message:

SIP/2.0 100 Trying
Via: SIP/2.0/WSS prlp9f6l80p0.invalid;branch=z9hG4bK7043746
From: "SIP.js Demo" <sip:200@10.50.100.73>;tag=1oe7kn4hoa
To: <sip:D121@{FQDN}>
Call-ID: g3rt9hcjhq5sn7nnlmd7
CSeq: 1 INVITE
Server: Brekeke SIP Server
Content-Length: 0

logger-factory.ts:109 Wed Mar 24 2021 19:15:55 GMT-0400 (Eastern Daylight Time) | sip.Inviter | Inviter.onTrying
logger-factory.ts:109 Wed Mar 24 2021 19:15:55 GMT-0400 (Eastern Daylight Time) | sip.Transport | Received WebSocket text message:

SIP/2.0 603 Decline
Via: SIP/2.0/WSS prlp9f6l80p0.invalid;branch=z9hG4bK7043746
From: "SIP.js Demo" <sip:200@10.50.100.73>;tag=1oe7kn4hoa
To: <sip:D121@{FQDN}>;tag=be0172dccs
Call-ID: g3rt9hcjhq5sn7nnlmd7
CSeq: 1 INVITE
Server: Brekeke SIP Server
Content-Length: 0

logger-factory.ts:109 Wed Mar 24 2021 19:15:55 GMT-0400 (Eastern Daylight Time) | sip.Transport | Sending WebSocket message:

ACK sip:D121@{FQDN} SIP/2.0
Via: SIP/2.0/WSS prlp9f6l80p0.invalid;branch=z9hG4bK7043746
To: <sip:D121@{FQDN}>;tag=be0172dccs
From: "SIP.js Demo" <sip:200@10.50.100.73>;tag=1oe7kn4hoa
Call-ID: g3rt9hcjhq5sn7nnlmd7
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0

Have you checked the wiki page below?
https://docs.brekeke.com/pbx/setting-up ... ser-webrtc

Also you need to use PBX's default DialPlan rules. If you use custom rules, disable them for testing.

Harold wrote:Have you checked the wiki page below?
https://docs.brekeke.com/pbx/setting-up ... ser-webrtc

Also you need to use PBX's default DialPlan rules. If you use custom rules, disable them for testing.

Hey Harold thanks for the reply,

I have followed the wiki. I have disabled all the dial plan rules. When you say default DialPlan are you referring to ARS?

Yes.
The default DialPlan rule which forwards a call to ARS is required to bridge WebRTC to/from non-WebRTC.

It sounds like there is a default DialPlan that is separate from ARS? Would this be under Options -> Settings?

Default DialPlan rules are listed under PBX's [Dial Plan]->[Rules] page.

If there are any non-default rules with higher priority, please disable them for testing.

I have [SIP SERVER] -> [Dial Plan] -> [Rules] in which I don't see any defaults and I have disabled all of them. Now I get a 404 Not Found with UDP and WebSockets.

Should add that someone did import the rules so its possible any defaults might have gotten removed.

You can download the default DialPlan table file for Brekeke PBX 3.10 from the link below.
https://brekeke-sip.com/bbs/dialplan/dialplan_3_10.tbl

I have replaced with the default Dial Plans and now I'm getting 407 Proxy Authentication Required. I was attempting to do with with out Authentication and had set Register, Invite to off

Any thoughts?

Thanks.

Have you restarted the PBX after you disabled Authentication settings?

Can you make a registration successfully without an Authentication error?

I was able to resolve the issue by setting out part of the bridge to force RTP.

Thanks, everyone for your help.