Audio and video call issue after call is getting connected in webrtc asterisk

Asterisk Asterisk WebRTC
1

Hi ,
I have configured webrtc in asterisk ,below its my configuration Calls were working properly suddenly it stopped working.I have added my configuration and issue logs in it.I have checked NAT confiugration also.

[401]
host=dynamic
secret=e2info
context=from-internal
type=friend
encryption=yes
avpf=yes
force_avp=yes
icesupport=yes
transport=wss,ws
directmedia=no
canreinvite=no
qualify_frequency = 60
disallow=all
allow=g729
allow=gsm
allow=g723
allow=ulaw
allow=vp8
allow=vp9
allow=h264
videosupport=yes
dtlsenable=yes
dtls_force_version_0 = false
dtlsverify=fingerprint
dtlscertfile=/etc/asterisk/keysc/wildcard_mycatie_com.crt
dtlsprivatekey=/etc/asterisk/keysc/wildcard_mycatie_com_Private.key
;tlscafile=/etc/asterisk/keysc/Chain.crt
dtlssetup=actpass
rtcp_mux=yes

issue logs
<------------->
— (8 headers 0 lines) —
Retransmitting #8 (NAT) to 84.17.43.186:50908:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 0.0.0.0:50908;branch=z9hG4bK1912656336;received=84.17.43.186;rport=50908
From: sip:00000000000113231228238:5060@3.231.228.238;tag=120083922
To: sip:0000000000011972592277524@3.231.228.238;tag=as095c6bdf
Call-ID: 868497745-403869607-2027512051
CSeq: 1 INVITE
Server: Asterisk PBX 13.38.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

Really destroying SIP dialog ‘6161b3cc623f2ae97561a6e51330b79c@3.231.228.238:0’ Method: INVITE
Really destroying SIP dialog ‘346l333t8276om4e4k0hj4’ Method: REGISTER
Retransmitting #9 (NAT) to 84.17.43.186:50908:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 0.0.0.0:50908;branch=z9hG4bK1912656336;received=84.17.43.186;rport=50908
From: sip:00000000000113231228238:5060@3.231.228.238;tag=120083922
To: sip:0000000000011972592277524@3.231.228.238;tag=as095c6bdf
Call-ID: 868497745-403869607-2027512051
CSeq: 1 INVITE
Server: Asterisk PBX 13.38.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

2

You haven’t included the whole transaction, which you need to do for a useful answer.

However, the response is being sent over UDP, and webrtc never uses UDP for signalling, so she “webrtc” in your subject cannot be true.

Failing to get an ACK, which is what is happening here, suggests you don’t have a usable route to 84.17.43.186, port 50908.

Are you using some sort of TLS to UDP gateway?

I suspect you are actually matching an allowguest setting, and using the default context. Complete logging would have made that clear.

3

This topic was automatically closed 30 days after the last reply. New replies are no longer allowed.