[Freeswitch-users] Call decline with 603

Wed Feb 18 17:28:13 MSK 2015

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 at 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 at 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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