[Kamailio-Users] BYE message not relayed to caller

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Vikram Ragukumar

17 Feb 2010 17 Feb '10

6:29 p.m.

Hello,

My setup includes a cellphone that uses Wi-Fi to hookup to the internet with IP 192.168.1.101. Cellphone app, tries to register with VOIPSwitch through an outbound proxy server(Kamailio + rtpproxy), which rewrites SDP to force rtp through an rtpproxy.

I can make a call from the cellphone to a PSTN number. When the PSTN endpoint hangs up, VOIPSwitch relays a BYE message to the proxy that looks like :

No. Time Source Destination Protocol Info 27 18.383929 IPaddr_VoipSwitch IPaddr_proxy SIP Request: BYE sip:1234@IPaddr_VoipSwitch

Frame 27 (420 bytes on wire, 420 bytes captured) Ethernet II, Src: Unispher_40:b5:39 (00:90:1a:40:b5:39), Dst: Supermic_bd:b9:bc (00:30:48:bd:b9:bc) Internet Protocol, Src: IPaddr_VoipSwitch , Dst: IPaddr_proxy User Datagram Protocol, Src Port: sip (5060), Dst Port: 7160 (7160) Session Initiation Protocol Request-Line: BYE sip:1234@IPaddr_VoipSwitch SIP/2.0 Method: BYE [Resent Packet: False] Message Header Route: sip:IPaddr_proxy:7160;lr=on;nat=yes CSeq: 1 BYE Via: SIP/2.0/UDP IPaddr_VoipSwitch:5060; branch=z9hG4bk180242100434182912333954 From: sip:5678@IPaddr_VoipSwitch;tag=18024210041929123187506289 Call-ID: LUOyIDm-ecy36NcUfj5G3v1lM6g5snkM To: "1234" sip:1234@IPaddr_VoipSwitch;tag=9XHJmiuf58vTrF7.vwflEK-G63JVPQSi Content-Length: 0

The request URI in the above message is the IP address of VOIPSwitch itself, which causes the proxy to forward the BYE message back to VOIPSwitch. The cellphone never receives the BYE message and consequently never hangs up.

Any suggestions on how i can get around this problem ?

Thanks and Regards, Vikram.

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Vikram Ragukumar

17 Feb 17 Feb

6:48 p.m.

Hello,

I also want to add that, this behavior is not seen while using Asterisk instead of VOIPSwitch. While using Asterisk, the BYE message has a request URI corresponding to the cellphone, so the proxy forwards the BYE message to the cellphone which causes the cellphone to hang up.

Thanks in advance for your help, Vikram.

...

Hello,

My setup includes a cellphone that uses Wi-Fi to hookup to the internet with IP 192.168.1.101. Cellphone app, tries to register with VOIPSwitch through an outbound proxy server(Kamailio + rtpproxy), which rewrites SDP to force rtp through an rtpproxy.

I can make a call from the cellphone to a PSTN number. When the PSTN endpoint hangs up, VOIPSwitch relays a BYE message to the proxy that looks like :

No. Time Source Destination Protocol Info 27 18.383929 IPaddr_VoipSwitch IPaddr_proxy SIP Request: BYE sip:1234@IPaddr_VoipSwitch

Frame 27 (420 bytes on wire, 420 bytes captured) Ethernet II, Src: Unispher_40:b5:39 (00:90:1a:40:b5:39), Dst: Supermic_bd:b9:bc (00:30:48:bd:b9:bc) Internet Protocol, Src: IPaddr_VoipSwitch , Dst: IPaddr_proxy User Datagram Protocol, Src Port: sip (5060), Dst Port: 7160 (7160) Session Initiation Protocol Request-Line: BYE sip:1234@IPaddr_VoipSwitch SIP/2.0 Method: BYE [Resent Packet: False] Message Header Route: sip:IPaddr_proxy:7160;lr=on;nat=yes CSeq: 1 BYE Via: SIP/2.0/UDP IPaddr_VoipSwitch:5060; branch=z9hG4bk180242100434182912333954 From: sip:5678@IPaddr_VoipSwitch;tag=18024210041929123187506289 Call-ID: LUOyIDm-ecy36NcUfj5G3v1lM6g5snkM To: "1234" sip:1234@IPaddr_VoipSwitch;tag=9XHJmiuf58vTrF7.vwflEK-G63JVPQSi Content-Length: 0

The request URI in the above message is the IP address of VOIPSwitch itself, which causes the proxy to forward the BYE message back to VOIPSwitch. The cellphone never receives the BYE message and consequently never hangs up.

Any suggestions on how i can get around this problem ?

Thanks and Regards, Vikram.

Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users

-- Thanks and Regards, Vikram Ragukumar.

Daniel-Constantin Mierla

19 Feb 19 Feb

1:50 a.m.

Hello,

can you post the entire call flow, from initial invite to to the bye. There is some mistake done somewhere in the routing elements. The sip trace will help to identify where.

Cheers, Daniel

On 02/18/2010 12:48 AM, Vikram Ragukumar wrote:

...

Hello,

I also want to add that, this behavior is not seen while using Asterisk instead of VOIPSwitch. While using Asterisk, the BYE message has a request URI corresponding to the cellphone, so the proxy forwards the BYE message to the cellphone which causes the cellphone to hang up.

Thanks in advance for your help, Vikram.

...
Hello,

My setup includes a cellphone that uses Wi-Fi to hookup to the internet with IP 192.168.1.101. Cellphone app, tries to register with VOIPSwitch through an outbound proxy server(Kamailio + rtpproxy), which rewrites SDP to force rtp through an rtpproxy.

I can make a call from the cellphone to a PSTN number. When the PSTN endpoint hangs up, VOIPSwitch relays a BYE message to the proxy that looks like :

No. Time Source Destination Protocol Info 27 18.383929 IPaddr_VoipSwitch IPaddr_proxy SIP Request: BYE sip:1234@IPaddr_VoipSwitch

Frame 27 (420 bytes on wire, 420 bytes captured) Ethernet II, Src: Unispher_40:b5:39 (00:90:1a:40:b5:39), Dst: Supermic_bd:b9:bc (00:30:48:bd:b9:bc) Internet Protocol, Src: IPaddr_VoipSwitch , Dst: IPaddr_proxy User Datagram Protocol, Src Port: sip (5060), Dst Port: 7160 (7160) Session Initiation Protocol Request-Line: BYE sip:1234@IPaddr_VoipSwitch SIP/2.0 Method: BYE [Resent Packet: False] Message Header Route:sip:IPaddr_proxy:7160;lr=on;nat=yes CSeq: 1 BYE Via: SIP/2.0/UDP IPaddr_VoipSwitch:5060; branch=z9hG4bk180242100434182912333954 From: sip:5678@IPaddr_VoipSwitch;tag=18024210041929123187506289 Call-ID: LUOyIDm-ecy36NcUfj5G3v1lM6g5snkM To: "1234" sip:1234@IPaddr_VoipSwitch;tag=9XHJmiuf58vTrF7.vwflEK-G63JVPQSi Content-Length: 0

The request URI in the above message is the IP address of VOIPSwitch itself, which causes the proxy to forward the BYE message back to VOIPSwitch. The cellphone never receives the BYE message and consequently never hangs up.

Any suggestions on how i can get around this problem ?

Thanks and Regards, Vikram.

Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users

-- Daniel-Constantin Mierla SIP Server Professional Solutions * http://www.asipto.com/

Vikram Ragukumar

22 Feb 22 Feb

12:10 p.m.

Daniel,

Thank you for your response. Please find below the sip trace from INVITE to BYE. These SIP messages were captured at the input interface of the proxy, so the SDP is not rewritten yet.

No. Time Source Destination Protocol Info 16 5.676114 Cell_phone_gw Proxy SIP/SDP Request: INVITE sip:1234@VoipSwitch:5060, with session description

Frame 16 (1264 bytes on wire, 1264 bytes captured) Ethernet II, Src: 00:26:f2:c8:44:11 (00:26:f2:c8:44:11), Dst: Supermic_bd:b9:bc (00:30:48:bd:b9:bc) Internet Protocol, Src: Cell_phone_gw (Cell_phone_gw), Dst: Proxy (Proxy) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: INVITE sip:1234@VoipSwitch:5060 SIP/2.0 Message Header Via: SIP/2.0/UDP 192.168.1.101:5060;rport;branch=z9hG4bKPjVuvvDqG5otxrgR6y9gyqnqWOpoBvIGXW Max-Forwards: 70 From: "91131" sip:91131@VoipSwitch;tag=TT8yqL1H.F8Pw9K1m.p38ry157WF5xdK To: sip:1234@VoipSwitch Contact: "91131" sip:91131@192.168.1.101:5060 Call-ID: s2iBR8MRCrDGTcqUah7j9RqDfbqDzefD CSeq: 24680 INVITE Route: sip:Proxy:5060;lr Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 Proxy-Authorization: Digest username="91131", realm="VoipSwitch", nonce="126686109922231105302513908108", uri="sip:1234@VoipSwitch:5060", response="55122bcb903503303164237e62481f52" Content-Type: application/sdp Content-Length: 379 Message Body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 3475932668 3475932668 IN IP4 192.168.1.101 Session Name (s): pjmedia Connection Information (c): IN IP4 192.168.1.101 Time Description, active time (t): 0 0 Session Attribute (a): X-nat:0 Media Description, name and address (m): audio 4000 RTP/AVP 114 18 113 0 8 101 Media Attribute (a): rtcp:4001 IN IP4 192.168.1.101 Media Attribute (a): rtpmap:114 AMR/8000 Media Attribute (a): rtpmap:18 G729/8000 Media Attribute (a): rtpmap:113 iLBC/8000 Media Attribute (a): fmtp:113 mode=30 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): sendrecv Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-15

No. Time Source Destination Protocol Info 17 5.744897 Proxy Cell_phone_gw SIP Status: 100 Giving a try

Frame 17 (429 bytes on wire, 429 bytes captured) Ethernet II, Src: Supermic_bd:b9:bc (00:30:48:bd:b9:bc), Dst: 00:26:f2:c8:44:11 (00:26:f2:c8:44:11) Internet Protocol, Src: Proxy (Proxy), Dst: Cell_phone_gw (Cell_phone_gw) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 100 Giving a try Message Header Via: SIP/2.0/UDP 192.168.1.101:5060;rport=5060;branch=z9hG4bKPjVuvvDqG5otxrgR6y9gyqnqWOpoBvIGXW;received=Cell_phone_gw From: "91131" sip:91131@VoipSwitch;tag=TT8yqL1H.F8Pw9K1m.p38ry157WF5xdK To: sip:1234@VoipSwitch Call-ID: s2iBR8MRCrDGTcqUah7j9RqDfbqDzefD CSeq: 24680 INVITE Server: Kamailio (1.5.3-notls (i386/linux)) Content-Length: 0

No. Time Source Destination Protocol Info 18 5.747037 Proxy VoipSwitch SIP/SDP Request: INVITE sip:1234@VoipSwitch:5060, with session description

Frame 18 (1434 bytes on wire, 1434 bytes captured) Ethernet II, Src: Supermic_bd:b9:bc (00:30:48:bd:b9:bc), Dst: Unispher_40:b5:39 (00:90:1a:40:b5:39) Internet Protocol, Src: Proxy (Proxy), Dst: VoipSwitch (VoipSwitch) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: INVITE sip:1234@VoipSwitch:5060 SIP/2.0 Message Header Record-Route: sip:Proxy:5060;lr=on;nat=yes Via: SIP/2.0/UDP Proxy:5060;branch=z9hG4bKd5eb.409fb37.0 Via: SIP/2.0/UDP 192.168.1.101:5060;received=Cell_phone_gw;rport=5060;branch=z9hG4bKPjVuvvDqG5otxrgR6y9gyqnqWOpoBvIGXW Max-Forwards: 69 From: "91131" sip:91131@VoipSwitch;tag=TT8yqL1H.F8Pw9K1m.p38ry157WF5xdK To: sip:1234@VoipSwitch Contact: "91131" sip:91131@Cell_phone_gw:5060 Call-ID: s2iBR8MRCrDGTcqUah7j9RqDfbqDzefD CSeq: 24680 INVITE Route: sip:Proxy:5060;lr Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 Proxy-Authorization: Digest username="91131", realm="VoipSwitch", nonce="126686109922231105302513908108", uri="sip:1234@VoipSwitch:5060", response="55122bcb903503303164237e62481f52" Content-Type: application/sdp Content-Length: 379 P-hint: outbound Message Body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 3475932668 3475932668 IN IP4 192.168.1.101 Session Name (s): pjmedia Connection Information (c): IN IP4 Proxy Time Description, active time (t): 0 0 Session Attribute (a): X-nat:0 Media Description, name and address (m): audio 35752 RTP/AVP 114 18 113 0 8 101 Media Attribute (a): rtcp:35753 Media Attribute (a): rtpmap:114 AMR/8000 Media Attribute (a): rtpmap:18 G729/8000 Media Attribute (a): rtpmap:113 iLBC/8000 Media Attribute (a): fmtp:113 mode=30 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): sendrecv Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-15 Media Attribute (a): nortpproxy:yes

No. Time Source Destination Protocol Info 19 5.934950 VoipSwitch Proxy SIP Status: 100 Trying

Frame 19 (579 bytes on wire, 579 bytes captured) Ethernet II, Src: Unispher_40:b5:39 (00:90:1a:40:b5:39), Dst: Supermic_bd:b9:bc (00:30:48:bd:b9:bc) Internet Protocol, Src: VoipSwitch (VoipSwitch), Dst: Proxy (Proxy) User Datagram Protocol, Src Port: sip (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 100 Trying Message Header CSeq: 24680 INVITE Via: SIP/2.0/UDP Proxy:5060;branch=z9hG4bKd5eb.409fb37.0 Via: SIP/2.0/UDP 192.168.1.101:5060;received=Cell_phone_gw;rport=5060;branch=z9hG4bKPjVuvvDqG5otxrgR6y9gyqnqWOpoBvIGXW From: "91131" sip:91131@VoipSwitch;tag=TT8yqL1H.F8Pw9K1m.p38ry157WF5xdK Call-ID: s2iBR8MRCrDGTcqUah7j9RqDfbqDzefD To: sip:1234@VoipSwitch;tag=22025110233933268788916305 Contact: sip:VoipSwitch:5060;transport=udp Content-Length: 0 Record-Route: sip:Proxy:5060;lr=on;nat=yes

No. Time Source Destination Protocol Info 20 6.707560 VoipSwitch Proxy SIP/SDP Status: 183 Session Progress, with session description

Frame 20 (868 bytes on wire, 868 bytes captured) Ethernet II, Src: Unispher_40:b5:39 (00:90:1a:40:b5:39), Dst: Supermic_bd:b9:bc (00:30:48:bd:b9:bc) Internet Protocol, Src: VoipSwitch (VoipSwitch), Dst: Proxy (Proxy) User Datagram Protocol, Src Port: sip (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 183 Session Progress Message Header CSeq: 24680 INVITE Via: SIP/2.0/UDP Proxy:5060;branch=z9hG4bKd5eb.409fb37.0 Via: SIP/2.0/UDP 192.168.1.101:5060;received=Cell_phone_gw;rport=5060;branch=z9hG4bKPjVuvvDqG5otxrgR6y9gyqnqWOpoBvIGXW From: "91131" sip:91131@VoipSwitch;tag=TT8yqL1H.F8Pw9K1m.p38ry157WF5xdK Call-ID: s2iBR8MRCrDGTcqUah7j9RqDfbqDzefD To: sip:1234@VoipSwitch;tag=22025110233933268788916305 Contact: sip:VoipSwitch:5060;transport=udp Content-Type: application/sdp Content-Length: 246 Record-Route: sip:Proxy:5060;lr=on;nat=yes Message Body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): VoipSwitch 7304 7304 IN IP4 VoipSwitch Session Name (s): VoipSIP Session Information (i): Audio Session Connection Information (c): IN IP4 VoipSwitch Time Description, active time (t): 0 0 Media Description, name and address (m): audio 6304 RTP/AVP 18 101 Media Attribute (a): rtpmap:18 G729/8000/1 Media Attribute (a): fmtp:18 annexb=no Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-15 Media Attribute (a): sendrecv

No. Time Source Destination Protocol Info 21 6.734267 Proxy Cell_phone_gw SIP/SDP Status: 183 Session Progress, with session description

Frame 21 (822 bytes on wire, 822 bytes captured) Ethernet II, Src: Supermic_bd:b9:bc (00:30:48:bd:b9:bc), Dst: 00:26:f2:c8:44:11 (00:26:f2:c8:44:11) Internet Protocol, Src: Proxy (Proxy), Dst: Cell_phone_gw (Cell_phone_gw) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 183 Session Progress Message Header CSeq: 24680 INVITE Via: SIP/2.0/UDP 192.168.1.101:5060;received=Cell_phone_gw;rport=5060;branch=z9hG4bKPjVuvvDqG5otxrgR6y9gyqnqWOpoBvIGXW From: "91131" sip:91131@VoipSwitch;tag=TT8yqL1H.F8Pw9K1m.p38ry157WF5xdK Call-ID: s2iBR8MRCrDGTcqUah7j9RqDfbqDzefD To: sip:1234@VoipSwitch;tag=22025110233933268788916305 Contact: sip:VoipSwitch:5060;transport=udp Content-Type: application/sdp Content-Length: 267 Record-Route: sip:Proxy:5060;lr=on;nat=yes Message Body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): VoipSwitch 7304 7304 IN IP4 VoipSwitch Session Name (s): VoipSIP Session Information (i): Audio Session Connection Information (c): IN IP4 Proxy Time Description, active time (t): 0 0 Media Description, name and address (m): audio 35570 RTP/AVP 18 101 Media Attribute (a): rtpmap:18 G729/8000/1 Media Attribute (a): fmtp:18 annexb=no Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-15 Media Attribute (a): sendrecv Media Attribute (a): nortpproxy:yes

No. Time Source Destination Protocol Info 22 15.889935 Cell_phone_gw Proxy UDP Source port: 5060 Destination port: 5060

Frame 22 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: 00:26:f2:c8:44:11 (00:26:f2:c8:44:11), Dst: Supermic_bd:b9:bc (00:30:48:bd:b9:bc) Internet Protocol, Src: Cell_phone_gw (Cell_phone_gw), Dst: Proxy (Proxy) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Data (2 bytes)

0000 0d 0a ..

No. Time Source Destination Protocol Info 23 19.801513 VoipSwitch Proxy SIP/SDP Status: 200 OK, with session description

Frame 23 (854 bytes on wire, 854 bytes captured) Ethernet II, Src: Unispher_40:b5:39 (00:90:1a:40:b5:39), Dst: Supermic_bd:b9:bc (00:30:48:bd:b9:bc) Internet Protocol, Src: VoipSwitch (VoipSwitch), Dst: Proxy (Proxy) User Datagram Protocol, Src Port: sip (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header CSeq: 24680 INVITE Via: SIP/2.0/UDP Proxy:5060;branch=z9hG4bKd5eb.409fb37.0 Via: SIP/2.0/UDP 192.168.1.101:5060;received=Cell_phone_gw;rport=5060;branch=z9hG4bKPjVuvvDqG5otxrgR6y9gyqnqWOpoBvIGXW From: "91131" sip:91131@VoipSwitch;tag=TT8yqL1H.F8Pw9K1m.p38ry157WF5xdK Call-ID: s2iBR8MRCrDGTcqUah7j9RqDfbqDzefD To: sip:1234@VoipSwitch;tag=22025110233933268788916305 Contact: sip:VoipSwitch:5060;transport=udp Content-Type: application/sdp Content-Length: 246 Record-Route: sip:Proxy:5060;lr=on;nat=yes Message Body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): VoipSwitch 7304 7304 IN IP4 VoipSwitch Session Name (s): VoipSIP Session Information (i): Audio Session Connection Information (c): IN IP4 VoipSwitch Time Description, active time (t): 0 0 Media Description, name and address (m): audio 6304 RTP/AVP 18 101 Media Attribute (a): rtpmap:18 G729/8000/1 Media Attribute (a): fmtp:18 annexb=no Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-15 Media Attribute (a): sendrecv

No. Time Source Destination Protocol Info 24 19.851387 Proxy Cell_phone_gw SIP/SDP Status: 200 OK, with session description

Frame 24 (808 bytes on wire, 808 bytes captured) Ethernet II, Src: Supermic_bd:b9:bc (00:30:48:bd:b9:bc), Dst: 00:26:f2:c8:44:11 (00:26:f2:c8:44:11) Internet Protocol, Src: Proxy (Proxy), Dst: Cell_phone_gw (Cell_phone_gw) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header CSeq: 24680 INVITE Via: SIP/2.0/UDP 192.168.1.101:5060;received=Cell_phone_gw;rport=5060;branch=z9hG4bKPjVuvvDqG5otxrgR6y9gyqnqWOpoBvIGXW From: "91131" sip:91131@VoipSwitch;tag=TT8yqL1H.F8Pw9K1m.p38ry157WF5xdK Call-ID: s2iBR8MRCrDGTcqUah7j9RqDfbqDzefD To: sip:1234@VoipSwitch;tag=22025110233933268788916305 Contact: sip:VoipSwitch:5060;transport=udp Content-Type: application/sdp Content-Length: 267 Record-Route: sip:Proxy:5060;lr=on;nat=yes Message Body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): VoipSwitch 7304 7304 IN IP4 VoipSwitch Session Name (s): VoipSIP Session Information (i): Audio Session Connection Information (c): IN IP4 Proxy Time Description, active time (t): 0 0 Media Description, name and address (m): audio 35570 RTP/AVP 18 101 Media Attribute (a): rtpmap:18 G729/8000/1 Media Attribute (a): fmtp:18 annexb=no Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-15 Media Attribute (a): sendrecv Media Attribute (a): nortpproxy:yes

No. Time Source Destination Protocol Info 25 19.860918 Cell_phone_gw Proxy SIP Request: ACK sip:VoipSwitch:5060;transport=udp

Frame 25 (470 bytes on wire, 470 bytes captured) Ethernet II, Src: 00:26:f2:c8:44:11 (00:26:f2:c8:44:11), Dst: Supermic_bd:b9:bc (00:30:48:bd:b9:bc) Internet Protocol, Src: Cell_phone_gw (Cell_phone_gw), Dst: Proxy (Proxy) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: ACK sip:VoipSwitch:5060;transport=udp SIP/2.0 Message Header Via: SIP/2.0/UDP 192.168.1.101:5060;rport;branch=z9hG4bKPj3R8VFolrcYnGiHZ65Foh4rL9rghVDXUW Max-Forwards: 70 From: "91131" sip:91131@VoipSwitch;tag=TT8yqL1H.F8Pw9K1m.p38ry157WF5xdK To: sip:1234@VoipSwitch;tag=22025110233933268788916305 Call-ID: s2iBR8MRCrDGTcqUah7j9RqDfbqDzefD CSeq: 24680 ACK Route: sip:Proxy:5060;lr;nat=yes Content-Length: 0

No. Time Source Destination Protocol Info 26 19.901346 Proxy VoipSwitch SIP Request: ACK sip:VoipSwitch:5060;transport=udp

Frame 26 (521 bytes on wire, 521 bytes captured) Ethernet II, Src: Supermic_bd:b9:bc (00:30:48:bd:b9:bc), Dst: Unispher_40:b5:39 (00:90:1a:40:b5:39) Internet Protocol, Src: Proxy (Proxy), Dst: VoipSwitch (VoipSwitch) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: ACK sip:VoipSwitch:5060;transport=udp SIP/2.0 Message Header Via: SIP/2.0/UDP Proxy:5060;branch=z9hG4bKd5eb.409fb37.2 Via: SIP/2.0/UDP 192.168.1.101:5060;received=Cell_phone_gw;rport=5060;branch=z9hG4bKPj3R8VFolrcYnGiHZ65Foh4rL9rghVDXUW Max-Forwards: 69 From: "91131" sip:91131@VoipSwitch;tag=TT8yqL1H.F8Pw9K1m.p38ry157WF5xdK To: sip:1234@VoipSwitch;tag=22025110233933268788916305 Call-ID: s2iBR8MRCrDGTcqUah7j9RqDfbqDzefD CSeq: 24680 ACK Content-Length: 0

No. Time Source Destination Protocol Info 27 27.987188 VoipSwitch Proxy SIP Request: BYE sip:91131@VoipSwitch

Frame 27 (420 bytes on wire, 420 bytes captured) Ethernet II, Src: Unispher_40:b5:39 (00:90:1a:40:b5:39), Dst: Supermic_bd:b9:bc (00:30:48:bd:b9:bc) Internet Protocol, Src: VoipSwitch (VoipSwitch), Dst: Proxy (Proxy) User Datagram Protocol, Src Port: sip (5060), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: BYE sip:91131@VoipSwitch SIP/2.0 Message Header Route: sip:Proxy:5060;lr=on;nat=yes CSeq: 1 BYE Via: SIP/2.0/UDP VoipSwitch:5060;branch=z9hG4bk220252102301223326901297 From: sip:1234@VoipSwitch;tag=22025110233933268788916305 Call-ID: s2iBR8MRCrDGTcqUah7j9RqDfbqDzefD To: "91131" sip:91131@VoipSwitch;tag=TT8yqL1H.F8Pw9K1m.p38ry157WF5xdK Content-Length: 0

No. Time Source Destination Protocol Info 28 28.211030 Proxy VoipSwitch SIP Request: BYE sip:91131@VoipSwitch

Frame 28 (490 bytes on wire, 490 bytes captured) Ethernet II, Src: Supermic_bd:b9:bc (00:30:48:bd:b9:bc), Dst: Unispher_40:b5:39 (00:90:1a:40:b5:39) Internet Protocol, Src: Proxy (Proxy), Dst: VoipSwitch (VoipSwitch) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: BYE sip:91131@VoipSwitch SIP/2.0 Message Header Max-Forwards: 10 CSeq: 1 BYE Via: SIP/2.0/UDP Proxy:5060;branch=z9hG4bKa0bd.f0181ba.0 Via: SIP/2.0/UDP VoipSwitch:5060;rport=5060;received=VoipSwitch;branch=z9hG4bk220252102301223326901297 From: sip:1234@VoipSwitch;tag=22025110233933268788916305 Call-ID: s2iBR8MRCrDGTcqUah7j9RqDfbqDzefD To: "91131" sip:91131@VoipSwitch;tag=TT8yqL1H.F8Pw9K1m.p38ry157WF5xdK Content-Length: 0

No. Time Source Destination Protocol Info 29 28.698172 Proxy VoipSwitch SIP Request: BYE sip:91131@VoipSwitch

Frame 29 (490 bytes on wire, 490 bytes captured) Ethernet II, Src: Supermic_bd:b9:bc (00:30:48:bd:b9:bc), Dst: Unispher_40:b5:39 (00:90:1a:40:b5:39) Internet Protocol, Src: Proxy (Proxy), Dst: VoipSwitch (VoipSwitch) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: BYE sip:91131@VoipSwitch SIP/2.0 Message Header Max-Forwards: 10 CSeq: 1 BYE Via: SIP/2.0/UDP Proxy:5060;branch=z9hG4bKa0bd.f0181ba.0 Via: SIP/2.0/UDP VoipSwitch:5060;rport=5060;received=VoipSwitch;branch=z9hG4bk220252102301223326901297 From: sip:1234@VoipSwitch;tag=22025110233933268788916305 Call-ID: s2iBR8MRCrDGTcqUah7j9RqDfbqDzefD To: "91131" sip:91131@VoipSwitch;tag=TT8yqL1H.F8Pw9K1m.p38ry157WF5xdK Content-Length: 0

No. Time Source Destination Protocol Info 30 29.698214 Proxy VoipSwitch SIP Request: BYE sip:91131@VoipSwitch

Frame 30 (490 bytes on wire, 490 bytes captured) Ethernet II, Src: Supermic_bd:b9:bc (00:30:48:bd:b9:bc), Dst: Unispher_40:b5:39 (00:90:1a:40:b5:39) Internet Protocol, Src: Proxy (Proxy), Dst: VoipSwitch (VoipSwitch) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: BYE sip:91131@VoipSwitch SIP/2.0 Message Header Max-Forwards: 10 CSeq: 1 BYE Via: SIP/2.0/UDP Proxy:5060;branch=z9hG4bKa0bd.f0181ba.0 Via: SIP/2.0/UDP VoipSwitch:5060;rport=5060;received=VoipSwitch;branch=z9hG4bk220252102301223326901297 From: sip:1234@VoipSwitch;tag=22025110233933268788916305 Call-ID: s2iBR8MRCrDGTcqUah7j9RqDfbqDzefD To: "91131" sip:91131@VoipSwitch;tag=TT8yqL1H.F8Pw9K1m.p38ry157WF5xdK Content-Length: 0

No. Time Source Destination Protocol Info 31 30.941201 Cell_phone_gw Proxy UDP Source port: 5060 Destination port: 5060

Frame 31 (60 bytes on wire, 60 bytes captured) Ethernet II, Src: 00:26:f2:c8:44:11 (00:26:f2:c8:44:11), Dst: Supermic_bd:b9:bc (00:30:48:bd:b9:bc) Internet Protocol, Src: Cell_phone_gw (Cell_phone_gw), Dst: Proxy (Proxy) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Data (2 bytes)

0000 0d 0a ..

No. Time Source Destination Protocol Info 32 31.699278 Proxy VoipSwitch SIP Request: BYE sip:91131@VoipSwitch

Frame 32 (490 bytes on wire, 490 bytes captured) Ethernet II, Src: Supermic_bd:b9:bc (00:30:48:bd:b9:bc), Dst: Unispher_40:b5:39 (00:90:1a:40:b5:39) Internet Protocol, Src: Proxy (Proxy), Dst: VoipSwitch (VoipSwitch) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: BYE sip:91131@VoipSwitch SIP/2.0 Message Header Max-Forwards: 10 CSeq: 1 BYE Via: SIP/2.0/UDP Proxy:5060;branch=z9hG4bKa0bd.f0181ba.0 Via: SIP/2.0/UDP VoipSwitch:5060;rport=5060;received=VoipSwitch;branch=z9hG4bk220252102301223326901297 From: sip:1234@VoipSwitch;tag=22025110233933268788916305 Call-ID: s2iBR8MRCrDGTcqUah7j9RqDfbqDzefD To: "91131" sip:91131@VoipSwitch;tag=TT8yqL1H.F8Pw9K1m.p38ry157WF5xdK Content-Length: 0

Thanks and Regards, Vikram.

...

Hello,

can you post the entire call flow, from initial invite to to the bye. There is some mistake done somewhere in the routing elements. The sip trace will help to identify where.

Cheers, Daniel

...
...
Hello,

My setup includes a cellphone that uses Wi-Fi to hookup to the internet with IP 192.168.1.101. Cellphone app, tries to register with VOIPSwitch through an outbound proxy server(Kamailio + rtpproxy), which rewrites SDP to force rtp through an rtpproxy.

I can make a call from the cellphone to a PSTN number. When the PSTN endpoint hangs up, VOIPSwitch relays a BYE message to the proxy that looks like :

No. Time Source Destination Protocol Info 27 18.383929 IPaddr_VoipSwitch IPaddr_proxy SIP Request: BYE sip:1234@IPaddr_VoipSwitch

Frame 27 (420 bytes on wire, 420 bytes captured) Ethernet II, Src: Unispher_40:b5:39 (00:90:1a:40:b5:39), Dst: Supermic_bd:b9:bc (00:30:48:bd:b9:bc) Internet Protocol, Src: IPaddr_VoipSwitch , Dst: IPaddr_proxy User Datagram Protocol, Src Port: sip (5060), Dst Port: 7160 (7160) Session Initiation Protocol Request-Line: BYE sip:1234@IPaddr_VoipSwitch SIP/2.0 Method: BYE [Resent Packet: False] Message Header Route:sip:IPaddr_proxy:7160;lr=on;nat=yes CSeq: 1 BYE Via: SIP/2.0/UDP IPaddr_VoipSwitch:5060; branch=z9hG4bk180242100434182912333954 From: sip:5678@IPaddr_VoipSwitch;tag=18024210041929123187506289 Call-ID: LUOyIDm-ecy36NcUfj5G3v1lM6g5snkM To: "1234" sip:1234@IPaddr_VoipSwitch;tag=9XHJmiuf58vTrF7.vwflEK-G63JVPQSi Content-Length: 0

The request URI in the above message is the IP address of VOIPSwitch itself, which causes the proxy to forward the BYE message back to VOIPSwitch. The cellphone never receives the BYE message and consequently never hangs up.

Any suggestions on how i can get around this problem ?

Thanks and Regards, Vikram.

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-- Daniel-Constantin Mierla SIP Server Professional Solutions

http://www.asipto.com/

-- Thanks and Regards, Vikram Ragukumar.

Vikram Ragukumar

24 Feb 24 Feb

2:40 p.m.

Daniel,

I have tried to summarize the SIP message flow below. I am also including the entire SIP trace at the end of this message.

Cell Phone Kamailio Phone B | | | |INVITE | | |------------->| | |100 Trying | | |<-------------| | | |INVITE | | |------------->| | |100 trying | | |<-------------| | |183SessionProg| | |<-------------| |183SessionProg| | |<-------------| | | | 200 OK | | 200 OK |<-------------| |<-------------| | | ACK | | |------------->| | | | ACK | | |------------->| |200 OK | | |<-------------| | | | BYE | | |<-------------|<- BYE,RURI=account@VoipSwitch | | BYE | | |------------->| | | BYE | | |------------->|

What might be causing VoipSwitch to send a BYE with RURI=account@VoipSwitch? As a result the BYE message never gets forwarded to the cellphone, and the proxy repeatedly sends BYE messages back to VoipSwitch.

Thanks in advance for your help. Regards, Vikram.

PS : Below is the SIP trace for the above call flow.

---------------------------------------------------------------------------- No. Time Source Destination Protocol Info 16 5.676114 Cell_phone_gw Proxy SIP/SDP Request: INVITE sip:1234@VoipSwitch:5060, with session description