[Kamailio-Users] BYE message not relayed to caller

Daniel-Constantin Mierla miconda at gmail.com
Thu Feb 25 08:34:17 CET 2010



On 02/24/2010 08:40 PM, Vikram Ragukumar wrote:
> 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 at VoipSwitch
>            |              |     BYE      |
>            |              |------------->|
>            |              |     BYE      |
>            |              |------------->|
>
>
> What might be causing VoipSwitch

Which one is the voipswitch in the diagram above? You refer to it but 
not included in diagram.

Daniel

>   to send a BYE with RURI=account at 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 at 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 at 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 at VoipSwitch>;tag=TT8yqL1H.F8Pw9K1m.p38ry157WF5xdK
>          To: sip:1234 at VoipSwitch
>          Contact: "91131"<sip:91131 at 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 at 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 at VoipSwitch>;tag=TT8yqL1H.F8Pw9K1m.p38ry157WF5xdK
>          To: sip:1234 at 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 at 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 at 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 at VoipSwitch>;tag=TT8yqL1H.F8Pw9K1m.p38ry157WF5xdK
>          To: sip:1234 at VoipSwitch
>          Contact: "91131"<sip:91131 at 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 at 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 at VoipSwitch>;tag=TT8yqL1H.F8Pw9K1m.p38ry157WF5xdK
>          Call-ID: s2iBR8MRCrDGTcqUah7j9RqDfbqDzefD
>          To: sip:1234 at 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 at VoipSwitch>;tag=TT8yqL1H.F8Pw9K1m.p38ry157WF5xdK
>          Call-ID: s2iBR8MRCrDGTcqUah7j9RqDfbqDzefD
>          To: sip:1234 at 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 at VoipSwitch>;tag=TT8yqL1H.F8Pw9K1m.p38ry157WF5xdK
>          Call-ID: s2iBR8MRCrDGTcqUah7j9RqDfbqDzefD
>          To: sip:1234 at 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 at VoipSwitch>;tag=TT8yqL1H.F8Pw9K1m.p38ry157WF5xdK
>          Call-ID: s2iBR8MRCrDGTcqUah7j9RqDfbqDzefD
>          To: sip:1234 at 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 at VoipSwitch>;tag=TT8yqL1H.F8Pw9K1m.p38ry157WF5xdK
>          Call-ID: s2iBR8MRCrDGTcqUah7j9RqDfbqDzefD
>          To: sip:1234 at 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 at VoipSwitch>;tag=TT8yqL1H.F8Pw9K1m.p38ry157WF5xdK
>          To: sip:1234 at 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 at VoipSwitch>;tag=TT8yqL1H.F8Pw9K1m.p38ry157WF5xdK
>          To: sip:1234 at 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 at 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 at 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 at VoipSwitch;tag=22025110233933268788916305
>          Call-ID: s2iBR8MRCrDGTcqUah7j9RqDfbqDzefD
>          To: "91131"
> <sip:91131 at VoipSwitch>;tag=TT8yqL1H.F8Pw9K1m.p38ry157WF5xdK
>          Content-Length: 0
>
> No.     Time        Source                Destination           Protocol Info
>       28 28.211030   Proxy        VoipSwitch          SIP      Request: BYE
> sip:91131 at 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 at 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 at VoipSwitch;tag=22025110233933268788916305
>          Call-ID: s2iBR8MRCrDGTcqUah7j9RqDfbqDzefD
>          To: "91131"
> <sip:91131 at VoipSwitch>;tag=TT8yqL1H.F8Pw9K1m.p38ry157WF5xdK
>          Content-Length: 0
>
> No.     Time        Source                Destination           Protocol Info
>       29 28.698172   Proxy        VoipSwitch          SIP      Request: BYE
> sip:91131 at 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 at 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 at VoipSwitch;tag=22025110233933268788916305
>          Call-ID: s2iBR8MRCrDGTcqUah7j9RqDfbqDzefD
>          To: "91131"
> <sip:91131 at VoipSwitch>;tag=TT8yqL1H.F8Pw9K1m.p38ry157WF5xdK
>          Content-Length: 0
>
> No.     Time        Source                Destination           Protocol Info
>       30 29.698214   Proxy        VoipSwitch          SIP      Request: BYE
> sip:91131 at 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 at 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 at VoipSwitch;tag=22025110233933268788916305
>          Call-ID: s2iBR8MRCrDGTcqUah7j9RqDfbqDzefD
>          To: "91131"
> <sip:91131 at 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 at 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 at 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 at VoipSwitch;tag=22025110233933268788916305
>          Call-ID: s2iBR8MRCrDGTcqUah7j9RqDfbqDzefD
>          To: "91131"
> <sip:91131 at VoipSwitch>;tag=TT8yqL1H.F8Pw9K1m.p38ry157WF5xdK
>          Content-Length: 0
> ----------------------------------------------------------------------------
>
>    
>>> 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
>>>        
>
>
>
> _______________________________________________
> Kamailio (OpenSER) - Users mailing list
> Users at 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
Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010
* http://www.asipto.com/index.php/sip-router-masterclass/




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