SIP call issue using JsSIP in asterisk

Asterisk Asterisk WebRTC
1

I have tried to do sip call using webrtc with asterisk as signalling server.
I have successfully established connection between two peers and also triggered dialplan successfully.
But the call drops immediately with the following error.

Scheduling destruction of SIP dialog ‘437cac925ec317ed2d360a3606e27d76@10.10.1.153:5160’ in 32000 ms (Method: INVITE)
** == Everyone is busy/congested at this time (1:0/0/1)**
** – Executing [601@from-webrtc:6] NoOp(“SIP/600-0000001e”, " 58") in new stack**
** – Executing [601@from-webrtc:7] Hangup(“SIP/600-0000001e”, “”) in new stack**
** == Spawn extension (from-webrtc, 601, 7) exited non-zero on ‘SIP/600-0000001e’**
Scheduling destruction of SIP dialog ‘6vd8gon4tq4c8vlvq69v’ in 32000 ms (Method: INVITE)

<— Reliably Transmitting (NAT) to 10.10.1.130:7615 —>
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/WS sk1m9t5d13ab.invalid;branch=z9hG4bK1866557;received=10.10.1.130;rport=7615
From: sip:600@10.10.1.153;tag=nk7jvgspir
To: sip:601@10.10.1.153;tag=as52010ee9
Call-ID: 6vd8gon4tq4c8vlvq69v
CSeq: 7079 INVITE
Server: FPBX-13.0.197.8(13.17.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 90;refresher=uas
Content-Length: 0

The main reason is SIP/2.0 488 Not Acceptable Here . I have searched for the sollution and most of them suggested to check codecs between two peers and it seems to same for both peers**( Codecs : (ulaw|alaw|gsm)** . Could anyone please help me to sort out this issue.

Thanks in advance.

2

There should be a reason for this, in front of this message. If you can’t see one, the debugging level isn’t high enough.

You have got an unavailable status, which will either be the result of not being able to send the INVITE at at all, or something in the intial exchange. For the latter, you need to enable and include the trace of the SIP transaction that failed.

3

You also need to provide the configuration. WebRTC requires specific configuration to work or the SDP negotiation will fail and a 488 will be sent. If you plan on deploying WebRTC you should also spend time learning aspects of this (ICE, STUN, TURN, DTLS) as they are what WebRTC is built on.

4

@david551 and @jcolp,
Thanks for the response.

sip.conf

[600]
host=dynamic
secret=600
context=from-webrtc
type=friend
encryption=yes
avpf=yes
icesupport=yes
transport=ws,wss,udp
directmedia=no
disallow=all
dial = SIP/600
allow=ulaw
allow=alaw
allow=sppex
allow=gsm
dtlsenable=yes
dtlsverify=fingerprint
dtlscertfile=/etc/asterisk/keys/asterisk.pem
dtlscafile=/etc/asterisk/keys/ca.crt
dtlssetup=actpass
nat=yes,force_rport,comedia

5

I wasn’t aware that webrtc was supported on chan_sip.

Even if it is, my knowledge of chan_sip predates it and I believe that jcolp has little experience of it. chan_sip is deprecated, so shouldn’t be used for new designs.

6

WebRTC does exist in chan_sip, but it’s not really maintained so stuff can/will change and break it probably.

A full SIP trace is needed to look into this.