[Kamailio-Users] BYE message not relayed to caller

Vikram Ragukumar vragukumar at signalogic.com
Mon Mar 1 21:47:59 CET 2010


Daniel,

Thank you for your response.

We have verified that it is indeed a bug with VoipSwitch. We uninstalled
the older version (v2.0.0965) of VoipSwitch that we were running and
replaced it with a newer version (v2.985), and the problem went away.

Once again, thank you very much for your assistance.

Regards,
Vikram.

>
>
> On 02/24/2010 08:43 PM, Vikram Ragukumar wrote:
>> Hello,
>>
>> In the Call flow diagram Phone B is to be read as VoipSwitch.
>>
>
> ignore previous email, I read that first and replied immediately ...
>
>  From the sip trace, the INVITE going to B has good record-route and
> contact header. Therefore looks to be a bug in voipswitch.
>
> Daniel
>
>> Regards,
>> Vikram.
>>
>>> 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 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/
>
>


-- 
Thanks and Regards,
Vikram Ragukumar.




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