what about the PSTN side? did you check that? did call ever pass to the PSTN side? might be the error came from PSTN side? On Wed, Feb 18, 2015 at 4:28 PM, Rahul MathuR <rahul.ultimate@gmail.com> wrote:
Hi all,
I was doing a POC of WebRTC based audio call to PSTN, routed via kamailio (for protocol translation & proxy) and FS as SIP server. I get 100 trying from FS but after that 603 decline with reason - "Reason: Q.850;cause=16;text="NORMAL_CLEARING""
Also, my phone didn't ring whereas, ITU T Q850 codes say, 16 NORMAL_CLEARING normal call clearing [Q.850] This cause indicates that the call is being cleared because one of the users involved in the call has requested that the call be cleared. Under normal situations, the source of this cause is not the network.
Could you please help me in resolving this issue.
The messages are as under -
1. INVITE from JsSIP to Kamailio
INVITE sip:00919650926333@<FreeSwitch_IP> SIP/2.0 Route: <sip:<Kamailio_IP>:10080;transport=ws;lr> Via: SIP/2.0/TCP amadf8lur89p.invalid;branch=z9hG4bK1158107 Max-Forwards: 69 To: <sip:00919650926333@<FreeSwitch_IP>> From: "55555" <sip:55555@<FreeSwitch_IP>>;tag=3dp7hdgg6j Call-ID: n41t8s01dnclcbodd3il CSeq: 352 INVITE Proxy-Authorization: Digest algorithm=MD5, username="55555", realm="<FreeSwitch_IP>", nonce="d0593b9e-b772-11e4-aeec-2965abc3007e", uri="sip:00919650926333@<FreeSwitch_IP>", response="12e4164f793098007d8bf09f7b50815f", qop=auth, cnonce="l02s3ec50q1g", nc=00000001 Contact: <sip:220bscel@amadf8lur89p.invalid;transport=ws;ob> Content-Type: application/sdp Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS Supported: ice,outbound User-Agent: JsSIP 0.6.4 Content-Length: 1769
v=0 o=- 9056480915531460217 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE audio a=msid-semantic: WMS dcrNk6emj9gTfynUdLaYmsTVbZwnKc9iEwCG m=audio 13228 RTP/SAVPF 111 103 104 9 0 8 106 105 13 126 c=IN IP4 59.178.158.4 a=rtcp:13228 IN IP4 59.178.158.4 a=candidate:2437072876 1 udp 2122260223 192.168.1.2 64540 typ host generation 0 a=candidate:2437072876 2 udp 2122260223 192.168.1.2 64540 typ host generation 0 a=candidate:3753982748 1 tcp 1518280447 192.168.1.2 0 typ host tcptype active generation 0 a=candidate:3753982748 2 tcp 1518280447 192.168.1.2 0 typ host tcptype active generation 0 a=candidate:941443129 1 udp 1686052607 59.178.158.4 13228 typ srflx raddr 192.168.1.2 rport 64540 generation 0 a=candidate:941443129 2 udp 1686052607 59.178.158.4 13228 typ srflx raddr 192.168.1.2 rport 64540 generation 0 a=ice-ufrag:9Yhi1W5j+XrHKQKQ a=ice-pwd:fVw9fkXXv1bteXnWh3B/694c a=ice-options:google-ice a=fingerprint:sha-256 C1:96:B0:69:7A:4C:D6:3B:DD:6C:4B:83:BF:F6:45:56:43:95:B4:46:E0:11:BF:AB:2A:42:8D:47:F7:DA:A8:66 a=setup:actpass a=mid:audio a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time a=sendrecv a=rtcp-mux a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:1011570965 cname:m/wi9qp3sxYVEQLV a=ssrc:1011570965 msid:dcrNk6emj9gTfynUdLaYmsTVbZwnKc9iEwCG 9fd4df2c-5c5d-4dec-a1a6-e181d5da35c4 a=ssrc:1011570965 mslabel:dcrNk6emj9gTfynUdLaYmsTVbZwnKc9iEwCG a=ssrc:1011570965 label:9fd4df2c-5c5d-4dec-a1a6-e181d5da35c4
2. INVITE from Kamailio to FS
INVITE sip:00919560509733@<FreeSwitch_IP> SIP/2.0 Record-Route: <sip:A4c7QkcuqY2hdQV9Y7p+J2A7spo+LvA=@<Kamailio_IP>:5090;r2=on;lr=on> Record-Route: <sip:A4c7QkcuqY2hdQV9Y7p+J2A7spo+LvA=@<Kamailio_IP>:10080;transport=ws;r2=on;lr=on> Via: SIP/2.0/UDP <Kamailio_IP>:5090;branch=z9hG4bKb937.6279f7ec5448a6c7867563e9b50fbe39.0 Via: SIP/2.0/TCP 4dcddrn8nrh8.invalid;rport=12016;received=59.178.154.62;branch=z9hG4bK4195437 Max-Forwards: 68 To: <sip:00919560509733@<FreeSwitch_IP>> From: "55555" <sip:55555@<FreeSwitch_IP>>;tag=i5so9j18ej Call-ID: pk2gvvb0qancp1ki8v5t CSeq: 5489 INVITE Proxy-Authorization: Digest algorithm=MD5, username="55555", realm="<FreeSwitch_IP>", nonce="85496252-b74d-11e4-ae0e-2965abc3007e", uri="sip:00919560509733@<FreeSwitch_IP>", response="a821425e292b05c4b9480cb66336cb93", qop=auth, cnonce="h8i58dtuuc0p", nc=00000001 Contact: <sip:g5uvomah@4dcddrn8nrh8.invalid ;transport=ws;ob;alias=59.178.154.62~12016~5> Content-Type: application/sdp Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS Supported: ice,outbound User-Agent: JsSIP 0.6.4 Content-Length: 782
v=0 o=- 4750671857217657811 2 IN IP4 <Kamailio_IP> s=- t=0 0 a=msid-semantic: WMS IMbls497PsJs5HOf2sY8de0vWcfW3Vjim2Te m=audio 32838 RTP/AVP 111 103 104 9 0 8 106 105 13 126 c=IN IP4 <Kamailio_IP> a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:2268746120 cname:Bk0e7lSPEkClAxNy a=ssrc:2268746120 msid:IMbls497PsJs5HOf2sY8de0vWcfW3Vjim2Te 291b0246-4d75-4347-8b2d-e8e42171a36e a=ssrc:2268746120 mslabel:IMbls497PsJs5HOf2sY8de0vWcfW3Vjim2Te a=ssrc:2268746120 label:291b0246-4d75-4347-8b2d-e8e42171a36e a=sendrecv a=rtcp:32839
3. 100 trying from FS to Kamailio
SIP/2.0 100 Trying Via: SIP/2.0/UDP <Kamailio_IP>:5090;branch=z9hG4bKb937.6279f7ec5448a6c7867563e9b50fbe39.0 Via: SIP/2.0/TCP 4dcddrn8nrh8.invalid;rport=12016;received=59.178.154.62;branch=z9hG4bK4195437 Record-Route: <sip:A4c7QkcuqY2hdQV9Y7p+J2A7spo+LvA=@<Kamailio_IP>:5090;r2=on;lr=on> Record-Route: <sip:A4c7QkcuqY2hdQV9Y7p+J2A7spo+LvA=@<Kamailio_IP>:10080;transport=ws;r2=on;lr=on> From: "55555" <sip:55555@<FreeSwitch_IP>>;tag=i5so9j18ej To: <sip:00919560509733@<FreeSwitch_IP>> Call-ID: pk2gvvb0qancp1ki8v5t CSeq: 5489 INVITE User-Agent: ASTPP Content-Length: 0
4. 603 from FS to Kamailio
SIP/2.0 603 Decline Via: SIP/2.0/UDP <Kamailio_IP>:5090;branch=z9hG4bKb937.6279f7ec5448a6c7867563e9b50fbe39.0 Via: SIP/2.0/TCP 4dcddrn8nrh8.invalid;rport=12016;received=59.178.154.62;branch=z9hG4bK4195437 Max-Forwards: 68 From: "55555" <sip:55555@<FreeSwitch_IP>>;tag=i5so9j18ej To: <sip:00919560509733@<FreeSwitch_IP>>;tag=y4D44m427U2Kj Call-ID: pk2gvvb0qancp1ki8v5t CSeq: 5489 INVITE User-Agent: ASTPP Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, NOTIFY Supported: path, replaces Allow-Events: talk, hold, conference, refer Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 Remote-Party-ID: "00919560509733" <sip:00919560509733@ <FreeSwitch_IP>>;party=calling;privacy=off;screen=no
-- Warm Regds. MathuRahul
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