[SR-Users] ACK not sent out

[Nagorny, Dimitry]

Hello List,

I have a setup of two WebRTC-Clients registered to kamailio. RegA calls RegB. Signalling itself for connection seems to work until the point RegA receives the 200Ok and wants to send the ACK, kamailio throws errors. Here are the 200Ok with the following ACK and the errors that kamailio writes:

200OK (from RegB to RegA, successful):
SIP/2.0 200 OK
Record-Route: <sip:zfHrSW4KTMuuxQYK+gVKH5MK+gUR1jc=@10.250.5.74:8083;transport=ws;lr=on;ftag=3bege11omg>
Via: SIP/2.0/WSS 192.0.2.1;rport=54839;received=10.250.5.17;branch=z9hG4bK415866
To: <sip:dnagorny at r5dev-kam1.robot5.de>;tag=ecmmlvc4ee
From: "Testaccount" <sip:4 at r5dev-kam1.robot5.de>;tag=3bege11omg
Call-ID: vu8rkkgki2uupuf2cqf4
CSeq: 2017 INVITE
Contact: <sip:l1jevtog at 192.0.2.59;alias=10.250.5.17~54838~6;alias=10.250.5.17~54838~6;transport=ws>
Supported: ice,replaces,outbound
Content-Type: application/sdp
Content-Length: 1001

v=0
o=mozilla...THIS_IS_SDPARTA-47.0 4652302965311172452 0 IN IP4 0.0.0.0
s=-
t=0 0
a=sendrecv
a=fingerprint:sha-256 1A:88:2D:6E:9F:82:14:E0:A2:DB:A7:16:56:00:2C:25:1B:D2:D1:EE:B2:D5:ED:6F:E1:7C:5C:B4:1A:30:9C:85
a=group:BUNDLE audio
a=ice-options:trickle
a=msid-semantic:WMS *
m=audio 30710 UDP/TLS/RTP/SAVPF 111
c=IN IP4 10.250.5.74
a=candidate:0 1 UDP 2122252543 10.250.5.17 52581 typ host
a=candidate:1 1 UDP 1686052863 87.158.246.174 49807 typ srflx raddr 10.250.5.17 rport 52581
a=end-of-candidates
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=fmtp:111 maxplaybackrate=48000;stereo=1
a=ice-pwd:4d722747e7d3b3caa360ebe9e646fb44
a=ice-ufrag:2b7e23b8
a=mid:audio
a=msid:{5cea7c38-cc3e-4bd2-9a3b-5d2381f0acf9} {0236e1cf-dfdc-4c94-8ce0-59f52a23ebb1}
a=rtpmap:111 opus/48000/2
a=setup:active
a=ssrc:565594393 cname:{7df4b0da-093c-4c2d-b154-0b860b43f575}
a=sendrecv
a=rtcp:30710
a=rtcp-mux
a=candidate:Y2XhN5oQNoH6nRph 1 UDP 2122252799 10.250.5.74 30710 typ host

ACK (from RegA to RegB, unsuccessfull):
ACK sip:l1jevtog at 192.0.2.59;alias=10.250.5.17~54838~6;transport=ws SIP/2.0
Route: <sip:zfHrSW4KTMuuxQYK+gVKH5MK+gUR1jc=@10.250.5.74:8083;transport=ws;lr=on;ftag=3bege11omg>
Via: SIP/2.0/WSS 192.0.2.1;branch=z9hG4bK2821443
Max-Forwards: 69
To: <sip:dnagorny at r5dev-kam1.robot5.de>;tag=ecmmlvc4ee
From: "Testaccount" <sip:4 at r5dev-kam1.robot5.de>;tag=3bege11omg
Call-ID: vu8rkkgki2uupuf2cqf4
CSeq: 2017 ACK
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
Supported: outbound
User-Agent: JsSIP 1.0.1
Content-Length: 0

Thrown Errors:
kamailio[5009]: INFO: <script>: Is natted
kamailio[5009]: INFO: <script>: From: sip:4 at r5dev-kam1.robot5.de To: sip:l1jevtog at 192.0.2.59;alias=10.250.5.17~54838~6;transport=ws @ <null> Via: SIP/2.0/WSS 192.0.2.1;branch=z9hG4bK4386608 Route: <null> vu8rkkgki2uupuf2cqf4
kamailio[5009]: ERROR: <core> [forward.c:529]: forward_request(): cannot forward to af 2, proto 5 no corresponding listening socket
kamailio[5009]: ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (7/SL)

I don't understand these errors. Can anybody explain and point me what needs to be fixed?
Using Kamailio 4.3.5 on CentOS7.2, funny info if I call from outside(PSTN/FreeSWITCH) to kamailio into WebRTC-Client it works w/o problems.


Respectfully
Dimitry Nagorny
Trainee

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