Re: [nist-sip] This SIP dialog don´t save all routin g information when I create an ACK Request?

"Jeroen van Bemmel" <[email protected]> Tue, 24 Apr 2007 20:27:21 +0200
Newsgroups gmane.comp.voip.nist-sip
Message-ID <006601c7869e$32854ce0$0601a8c0@BEMBUSTER>
Robert,

The ACK should contain Route headers containing all the URIs from the R-R 
headers. I quickly tested using the forkedinvite test in the TCK, when I add 
a R-R header twice it gets added twice to the ACK.

Please check that you have the latest sources, and compare your source code 
to that example

Regards,
Jeroen
  ----- Original Message ----- 
  From: Robert Andersson
  To: [email protected]
  Sent: Monday, April 23, 2007 4:19 PM
  Subject: [nist-sip] This SIP dialog don´t save all routing information 
when I create an ACK Request?


  Hello all.

  I have a question regarding the SIP dialog and the record-route and route 
header.

  Please look at the trace below. This is an outgoing call from my sip 
client to another client. When I receive the Response 200 OK I get 3 
addresses in the record route header field.

  Only the last one is inserted in the route filed of the ACK request? Do I 
miss a setting in the stack or is this an error?

  I set the javax.sip.OUTBOUND_PROXY parameter to 194.103.180.15.



  13:59:00,268 [SIPUserCls] DEBUG - ********************* SENDING REQUEST 
INVITE (OutGoing Call) **************************

  INVITE sip:[email protected] SIP/2.0

  Call-ID: [email protected]

  CSeq: 1 INVITE

  From: <sip:[email protected]>;tag=8691-King-7411

  To: <sip:[email protected]>

  Via: SIP/2.0/UDP 
194.103.180.10:5060;branch=z9hG4bK8816314e259046f7fa059c32189d6c14

  Max-Forwards: 70

  Content-Type: application/sdp

  Contact: <sip:[email protected]:5060>

  User-Agent: Netwise SIPgateway version 3.7.1.7 b1

  Content-Length: 168



  v=0

  o=Attendant 0 0 IN IP4 10.10.205.10

  s=SIP call

  c=IN IP4 10.10.205.10

  b=CT:256

  t=0 0

  m=video 25000 RTP/AVP 34

  a=rtpmap:34 H263/90000

  a=fmtp:34 CIF=1 QCIF=1





  13:59:00,344 [Gateway] DEBUG - ********************* SIP GATEWAY RESPONSE 
100 Trying ***************************

  SIP/2.0 100 Trying

  Call-ID: [email protected]

  CSeq: 1 INVITE

  From: <sip:[email protected]>;tag=8691-King-7411

  Via: SIP/2.0/UDP 
194.103.180.10:5060;branch=z9hG4bK8816314e259046f7fa059c32189d6c14

  Server: SIParator/4.2.3

  To: <sip:[email protected]>

  Content-Length: 0







  13:59:00,590 [Gateway] DEBUG - ********************* SIP GATEWAY RESPONSE 
180 Ringing ***************************

  SIP/2.0 180 Ringing

  From: <sip:[email protected]>;tag=8691-King-7411

  To: 
<sip:[email protected]>;tag=2356e18-ae00a8c0-13c4-2181-5040f23d-2181

  Call-ID: [email protected]

  CSeq: 1 INVITE

  Via: SIP/2.0/UDP 
194.103.180.10:5060;branch=z9hG4bK8816314e259046f7fa059c32189d6c14

  Supported: 100rel,replaces

  Record-Route: 
<sip:eDHgoGZDwuOqAm3nYAqBPuBaLE69Peceg62qNnLppvGI-pa0__h-9A-MF-PHr6mod@194.103.180.15;lr>,<sip:eSNow3Hh58mfBbdNrjtHox60RByIttSq521tSs1d30e7Aw35qUgDbA6z255EPOIqE5WddxQ0cWatYf1zT-a4p5g..@194.103.180.15;lr>,<sip:[email protected];lr>

  Content-Length: 0









  13:59:32,163 [Gateway] DEBUG - ********************* SIP GATEWAY RESPONSE 
200 OK ***************************

  SIP/2.0 200 OK

  From: <sip:[email protected]>;tag=8691-King-7411

  To: 
<sip:[email protected]>;tag=2356e18-ae00a8c0-13c4-2181-5040f23d-2181

  Call-ID: [email protected]

  CSeq: 1 INVITE

  Via: SIP/2.0/UDP 
194.103.180.10:5060;branch=z9hG4bK8816314e259046f7fa059c32189d6c14

  Supported: 100rel,replaces

  User-Agent: FranceTelecom/eConf

  Accept: 
application/sdp,audio/telephone-event,text/plain,text/html,application/media_control+xml,application/mc+xml,application/dtmf-relay,message/sipfrag

  Contact: 
<sip:ewqlOm8j7EmdRWErObv4yCYFZK0VCzCJslBso_S8EYa-OtFwsNvXxkwH5g23wxHCS@194.103.180.15>

  Record-Route: 
<sip:epfQBYlyW1jUwdtWtyFgbH78H9s2UJrU99eCBYE0NGZITSxKiXqAOVZObbAzfpnyX@194.103.180.15;lr>,<sip:evojwxngWoP5lmWvZmJq8p816ijVC0cCsSa-wn1QKac1c0gjqEyKm8MV65H6n8nacQ_npRc0UIEoIv9RLCRocvQ..@194.103.180.15;lr>,<sip:[email protected];lr>

  Allow: 
INVITE,ACK,BYE,CANCEL,OPTIONS,REFER,PRACK,INFO,MESSAGE,SUBSCRIBE,NOTIFY

  Content-Type: application/SDP

  Content-Length: 233



  v=0

  o=anonymous 1176465448 1176465417 IN IP4 194.103.180.15

  s=-

  i=eConf 4.1

  c=IN IP4 194.103.180.15

  b=AS:384

  t=0 0

  m=video 58300 RTP/AVP 34

  b=AS:352

  a=rtpmap:34 H263/90000

  a=TIAS:352000

  a=fmtp:34 CIF=1;QCIF=1

  a=sendrecv



  13:59:32,289 [OutgoingCall] DEBUG - ********************* SENDING REQUEST 
ACK (OutGoing Call) **************************

  ACK 
sip:ewqlOm8j7EmdRWErObv4yCYFZK0VCzCJslBso_S8EYa-OtFwsNvXxkwH5g23wxHCS@194.103.180.15 
SIP/2.0

  Call-ID: [email protected]

  CSeq: 1 ACK

  From: <sip:[email protected]>;tag=8691-King-7411

  To: 
<sip:[email protected]>;tag=2356e18-ae00a8c0-13c4-2181-5040f23d-2181

  Via: SIP/2.0/UDP 
194.103.180.10:5060;branch=z9hG4bK8816314e259046f7fa059c32189d6c14

  Max-Forwards: 70

  User-Agent: Netwise SIPgateway version 3.7.1.7 b1

  Route: <sip:[email protected];lr>

  Content-Length: 0





  Thanks for any help.



  Regards



  / Robert



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