HI Bogdan,
I'm already using an almsot identical version of uas.xml and uac.xml ( yes rrs=true) is being used. However in your version the uas.xml doesn't have rrs="true" after initial invite which I think is needed. See as you can see below, setting rrs="true" for uac will only work if it receives a Record-Route header in the 200OK which it's not.
In this case, ALL messages from openser to sipp uac do not contain the Record-route header. So I don't think it's a sipp problem, but an openser configuration problem. I've tried using other devices for a uac, such as x-lite but the same problem.
Andy
On 2/22/07, Bogdan-Andrei Iancu bogdan@voice-system.ro wrote:
Hi Andy,
so it's about sipp :D - I remember I had some hard times to make it work with record Route.
take a look at the attached files, they might help you.
regards, bogdan
Andy Pyles wrote:
HI Bogdan,
thanks for your reply. yes you are correct. The Bye doesn't have the Route header. It appears the the 200 OK sent to the caller doesn't contain a Record-route header. Messages between openser and callee contain record-route information, but messages between caller and openser do not. Is there a way to enable that?
Here's more detail: 192.168.0.101 = Caller (sipp) 1.2.3.4 = openser 4.3.2.1 = callee ( sipp)
1.) 192.168.0.101 -> 1.2.3.4 SIP/SDP Request: INVITE sip:service@1.2.3.4:5060, with session description 2.) 1.2.3.4 -> 192.168.0.101 SIP Status: 100 Giving a try 3.) 1.2.3.4 -> 4.3.2.1 SIP/SDP Request: INVITE sip:service@4.3.2.1:5060, with session description 4.) 4.3.2.1 -> 1.2.3.4 SIP Status: 180 Ringing 5.) 4.3.2.1 -> 1.2.3.4 SIP/SDP Status: 200 OK, with session description 6.) 1.2.3.4 -> 192.168.0.101 SIP Status: 180 Ringing 7.) 1.2.3.4 -> 192.168.0.101 SIP/SDP Status: 200 OK, with session description 8.) 192.168.0.101 -> 1.2.3.4 SIP Request: ACK sip:service@1.2.3.4:5060 9.) 1.2.3.4 -> 4.3.2.1 SIP Request: ACK sip:service@4.3.2.1:5060 10.) 192.168.0.101 -> 1.2.3.4 SIP Request: BYE sip:service@1.2.3.4:5060 11.) 1.2.3.4 -> 4.3.2.1 SIP Request: BYE sip:service@4.3.2.1:5060 12.) 4.3.2.1 -> 1.2.3.4 SIP Status: 200 OK 13.) 1.2.3.4 -> 192.168.0.101 SIP Status: 200 OK
Packets 6,7 and following contain no Record-route information. The other weird thing is that openser is passing on the Route: header it recevied from callee to the caller.
Please see attached for complete ngrep output.
On 2/21/07, Bogdan-Andrei Iancu bogdan@voice-system.ro wrote:
Hi Andy,
could you check on the net if the BYE contain the Route hdr added to INVITE as Record-Route? I have some doubts on this as I see: 0(966) find_first_route: No Route headers found 0(966) loose_route: There is no Route HF
and if the BYE is not identified, the dialog is not closed.
regards, bogdan
Andy Pyles wrote:
Hello,
I have a question on how to configure the dialog module ( 1.2.x from cvs yesterday ).
With my config, ( attached) I can make calls and have verified that the acc module is working correctly.
My question is, when I enable the dialog module, I can see that it is incrementing call count correctly, but when a bye is received, the dialog:active_dialogs statistic is never decremented.
In the debug level 9 logs, ( also attached) I see this error after the 200OK is sent to the bye:
1(969) DBUG:dialog:unref_dlg: unref dlg 0xa7ce5a98 with 1
(delete=0)-> 1
Is this a case of one of the timers being set too short? by the way using a variable call length from well under a second ( using sipp ) to 20 second call doesnt' seem to make a difference .
Thanks, Andy
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interface: lo (127.0.0.0/255.0.0.0) filter: (ip or ip6) and ( port 5060 ) # U 192.168.0.101:5060 -> 1.2.3.4:5060 INVITE sip:service@1.2.3.4:5060 SIP/2.0. Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK-1-0. From: sipp sip:sipp@192.168.0.101:5060;tag=22779SIPpTag001. To: sut sip:service@1.2.3.4:5060. Call-ID: 1-22779@192.168.0.101. CSeq: 1 INVITE. Contact: sip:sipp@192.168.0.101:5060. Max-Forwards: 70. Subject: Performance Test. Content-Type: application/sdp. Content-Length: 137. . v=0. o=user1 53655765 2353687637 IN IP4 192.168.0.101. s=-. c=IN IP4 192.168.0.101. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000.
# U 1.2.3.4:5060 -> 192.168.0.101:5060 SIP/2.0 100 Giving a try. Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK-1-0. From: sipp sip:sipp@192.168.0.101:5060;tag=22779SIPpTag001. To: sut sip:service@1.2.3.4:5060. Call-ID: 1-22779@192.168.0.101. CSeq: 1 INVITE. Server: OpenSer (1.2.0-pre6-notls (i386/linux)). Content-Length: 0. Warning: 392 1.2.3.4:5060 "Noisy feedback tells: pid=22746 req_src_ip=192.168.0.101 req_src_port=5060 in_uri=sip:service@1.2.3.4:5060 out_uri=sip:service@4.3.2.1:5060 via_cnt==1". .
# U 1.2.3.4:5060 -> 4.3.2.1:5060 INVITE sip:service@4.3.2.1:5060 SIP/2.0. Record-Route: sip:1.2.3.4;lr=on;ftag=22779SIPpTag001;did=052.8857fe74. Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bKa6e8.b1ea5ac3.0. Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK-1-0. From: sipp sip:sipp@192.168.0.101:5060;tag=22779SIPpTag001. To: sut sip:service@1.2.3.4:5060. Call-ID: 1-22779@192.168.0.101. CSeq: 1 INVITE. Contact: sip:sipp@192.168.0.101:5060. Max-Forwards: 69. Subject: Performance Test. Content-Type: application/sdp. Content-Length: 137. . v=0. o=user1 53655765 2353687637 IN IP4 192.168.0.101. s=-. c=IN IP4 192.168.0.101. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000.
# U 4.3.2.1:5060 -> 1.2.3.4:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bKa6e8.b1ea5ac3.0. Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK-1-0. From: sipp sip:sipp@192.168.0.101:5060;tag=22779SIPpTag001. To: sut sip:service@1.2.3.4:5060;tag=22700SIPpTag014. Call-ID: 1-22779@192.168.0.101. CSeq: 1 INVITE. Route: sip:1.2.3.4;lr=on;ftag=22779SIPpTag001;did=052.8857fe74. Contact: sip:4.3.2.1:5060;transport=UDP. Content-Length: 0. .
# U 4.3.2.1:5060 -> 1.2.3.4:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bKa6e8.b1ea5ac3.0. Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK-1-0. From: sipp sip:sipp@192.168.0.101:5060;tag=22779SIPpTag001. To: sut sip:service@1.2.3.4:5060;tag=22700SIPpTag014. Call-ID: 1-22779@192.168.0.101. CSeq: 1 INVITE. Route: sip:1.2.3.4;lr=on;ftag=22779SIPpTag001;did=052.8857fe74. Contact: sip:4.3.2.1:5060;transport=UDP. Content-Type: application/sdp. Content-Length: 125. . v=0. o=user1 53655765 2353687637 IN IP4 4.3.2.1. s=-. c=IN IP4 4.3.2.1. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000.
# U 1.2.3.4:5060 -> 192.168.0.101:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK-1-0. From: sipp sip:sipp@192.168.0.101:5060;tag=22779SIPpTag001. To: sut sip:service@1.2.3.4:5060;tag=22700SIPpTag014. Call-ID: 1-22779@192.168.0.101. CSeq: 1 INVITE. Route: sip:1.2.3.4;lr=on;ftag=22779SIPpTag001;did=052.8857fe74. Contact: sip:4.3.2.1:5060;transport=UDP. Content-Length: 0. .
# U 1.2.3.4:5060 -> 192.168.0.101:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK-1-0. From: sipp sip:sipp@192.168.0.101:5060;tag=22779SIPpTag001. To: sut sip:service@1.2.3.4:5060;tag=22700SIPpTag014. Call-ID: 1-22779@192.168.0.101. CSeq: 1 INVITE. Route: sip:1.2.3.4;lr=on;ftag=22779SIPpTag001;did=052.8857fe74. Contact: sip:4.3.2.1:5060;transport=UDP. Content-Type: application/sdp. Content-Length: 125. . v=0. o=user1 53655765 2353687637 IN IP4 4.3.2.1. s=-. c=IN IP4 4.3.2.1. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000.
# U 192.168.0.101:5060 -> 1.2.3.4:5060 ACK sip:service@1.2.3.4:5060 SIP/2.0. Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK-1-5. From: sipp sip:sipp@192.168.0.101:5060;tag=22779SIPpTag001. To: sut sip:service@1.2.3.4:5060;tag=22700SIPpTag014. Call-ID: 1-22779@192.168.0.101. CSeq: 1 ACK. Contact: sip:sipp@192.168.0.101:5060. Max-Forwards: 70. Subject: Performance Test. Content-Length: 0. .
# U 1.2.3.4:5060 -> 4.3.2.1:5060 ACK sip:service@4.3.2.1:5060 SIP/2.0. Record-Route: sip:1.2.3.4;lr=on;ftag=22779SIPpTag001. Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bKa6e8.b1ea5ac3.2. Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK-1-5. From: sipp sip:sipp@192.168.0.101:5060;tag=22779SIPpTag001. To: sut sip:service@1.2.3.4:5060;tag=22700SIPpTag014. Call-ID: 1-22779@192.168.0.101. CSeq: 1 ACK. Contact: sip:sipp@192.168.0.101:5060. Max-Forwards: 69. Subject: Performance Test. Content-Length: 0. .
# U 192.168.0.101:5060 -> 1.2.3.4:5060 BYE sip:service@1.2.3.4:5060 SIP/2.0. Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK-1-7. From: sipp sip:sipp@192.168.0.101:5060;tag=22779SIPpTag001. To: sut sip:service@1.2.3.4:5060;tag=22700SIPpTag014. Call-ID: 1-22779@192.168.0.101. CSeq: 2 BYE. Contact: sip:sipp@192.168.0.101:5060. Max-Forwards: 70. Subject: Performance Test. Content-Length: 0. .
# U 1.2.3.4:5060 -> 4.3.2.1:5060 BYE sip:service@4.3.2.1:5060 SIP/2.0. Record-Route: sip:1.2.3.4;lr=on;ftag=22779SIPpTag001. Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bK76e8.ecad2904.0. Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK-1-7. From: sipp sip:sipp@192.168.0.101:5060;tag=22779SIPpTag001. To: sut sip:service@1.2.3.4:5060;tag=22700SIPpTag014. Call-ID: 1-22779@192.168.0.101. CSeq: 2 BYE. Contact: sip:sipp@192.168.0.101:5060. Max-Forwards: 69. Subject: Performance Test. Content-Length: 0. .
# U 4.3.2.1:5060 -> 1.2.3.4:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bK76e8.ecad2904.0. Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK-1-7. From: sipp sip:sipp@192.168.0.101:5060;tag=22779SIPpTag001. To: sut sip:service@1.2.3.4:5060;tag=22700SIPpTag014. Call-ID: 1-22779@192.168.0.101. CSeq: 2 BYE. Route: sip:1.2.3.4;lr=on;ftag=22779SIPpTag001;did=052.8857fe74. Contact: sip:4.3.2.1:5060;transport=UDP. Content-Length: 0. .
# U 1.2.3.4:5060 -> 192.168.0.101:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK-1-7. From: sipp sip:sipp@192.168.0.101:5060;tag=22779SIPpTag001. To: sut sip:service@1.2.3.4:5060;tag=22700SIPpTag014. Call-ID: 1-22779@192.168.0.101. CSeq: 2 BYE. Route: sip:1.2.3.4;lr=on;ftag=22779SIPpTag001;did=052.8857fe74. Contact: sip:4.3.2.1:5060;transport=UDP. Content-Length: 0. .
exit 39 received, 0 dropped