7 Jul
2005
7 Jul
'05
7:05 a.m.
Hi everybody!
recently I have been testing calls to/from ISDN with a AS5350. My SER
version is 0.8.14 and when I make a call from ISDN towards a SIP UA
registered in the SER the problem is that the AS5350 is no aware of Call
Release.
I explain it in more detail...
1. AS5350 sends an INVITE to the SER with destination number 93222222
2. SER forwards the INVITE to the corresponding SIP UA.
3. UA reponds with a 180 Ringing and it is forwarded to the AS5350 by the
SER
4. When I pick up the phone, the SIP UA sends a 200 OK message, which is
frowarded by the SER as well.
5. AS5350 replies with a ACK to the 200 OK but beware!! the AS5350 has
changed UDP source port!! it is different from the one set in INVITE.
6. The SIP UA hangs up the call by sending a BYE message to the SER (it is
Record Routing)
7. The SER forwards the BYE to the AS5350 but with the old UDP port!!
8. AS5350 sends ICMP Destination Unreachable (Cause code: port unreachable)
every time that the SER retries the BYE.
My guess is that the AS5350 expects the BYE packet through the new UDP
port...
Has anybody found such a problem?
Thanks
Victor
7 Jul
7 Jul
8:18 a.m.
Hola Victor
Can you send the trace ?
ngrep at SER or debug ccsip messages at Cisco, and show sip-ua status.
----- Original Message -----
From: "Victor Huertas Garcia" <vhuertas(a)hotmail.com>
To: <serusers(a)lists.iptel.org>
Sent: Thursday, July 07, 2005 8:05 AM
Subject: [Serusers] URGENT! AS5350 <-> SER incompatibility problem
|
|
| Hi everybody!
|
|
| recently I have been testing calls to/from ISDN with a AS5350. My SER
| version is 0.8.14 and when I make a call from ISDN towards a SIP UA
| registered in the SER the problem is that the AS5350 is no aware of Call
| Release.
|
| I explain it in more detail...
|
| 1. AS5350 sends an INVITE to the SER with destination number 93222222
| 2. SER forwards the INVITE to the corresponding SIP UA.
| 3. UA reponds with a 180 Ringing and it is forwarded to the AS5350 by the
| SER
| 4. When I pick up the phone, the SIP UA sends a 200 OK message, which is
| frowarded by the SER as well.
| 5. AS5350 replies with a ACK to the 200 OK but beware!! the AS5350 has
| changed UDP source port!! it is different from the one set in INVITE.
| 6. The SIP UA hangs up the call by sending a BYE message to the SER (it is
| Record Routing)
| 7. The SER forwards the BYE to the AS5350 but with the old UDP port!!
| 8. AS5350 sends ICMP Destination Unreachable (Cause code: port unreachable)
| every time that the SER retries the BYE.
|
|
| My guess is that the AS5350 expects the BYE packet through the new UDP
| port...
|
| Has anybody found such a problem?
|
| Thanks
|
| Victor
|
|
| _______________________________________________
| Serusers mailing list
| serusers(a)lists.iptel.org
| http://lists.iptel.org/mailman/listinfo/serusers
9:06 a.m.
Here find the traces "debug ccsip messages"
*Jul 7 12:48:54.936: Sent:
INVITE sip:932222222@10.1.2.3:5060 SIP/2.0
Via: SIP/2.0/UDP 172.22.22.5:5060
From: <sip:172.22.22.5>;tag=25433214-696
To: <sip:932222222@10.1.2.3>
Date: Thu, 07 Jul 2005 12:48:54 GMT
Call-ID: 4FCAC87E-EE1C11D9-808AE1DB-AA3EFB56(a)172.22.22.5
Supported: timer,100rel
Min-SE: 1800
Cisco-Guid: 1338609710-3994816985-2156388827-2856254294
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE,
NOTIFY, INFO
CSeq: 101 INVITE
Max-Forwards: 6
Remote-Party-ID: <sip:172.22.22.5>;party=calling;screen=no;privacy=off
Timestamp: 1120740534
Contact: <sip:172.22.22.5:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 292
v=0
o=CiscoSystemsSIP-GW-UserAgent 362 6669 IN IP4 172.22.22.5
s=SIP Call
c=IN IP4 172.22.22.5
t=0 0
m=audio 17808 RTP/AVP 18 4 0 8
c=IN IP4 172.22.22.5
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=yes
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
*Jul 7 12:48:54.936: Received:
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 172.22.22.5:5060;rport=52053
From: <sip:172.22.22.5>;tag=25433214-696
To: <sip:932222222@10.1.2.3>
Call-ID: 4FCAC87E-EE1C11D9-808AE1DB-AA3EFB56(a)172.22.22.5
CSeq: 101 INVITE
Server: Sip EXpress router (0.8.14 (i386/linux))
Content-Length: 0
Warning: 392 10.1.2.3:5060 "Noisy feedback tells: pid=24175
req_src_ip=172.22.22.5 req_src_port=52053 in_uri=sip:932222222@10.1.2.3:5060
out_uri=sip:932222222@172.22.32.1:6000 via_cnt==1"
*Jul 7 12:48:55.692: Received:
SIP/2.0 180 Ringing
To: <sip:932222222@10.1.2.3>;tag=c525a909
From: <sip:172.22.22.5>;tag=25433214-696
Via: SIP/2.0/UDP 172.22.22.5:5060;rport=52053
Call-ID: 4FCAC87E-EE1C11D9-808AE1DB-AA3EFB56(a)172.22.22.5
CSeq: 101 INVITE
Record-Route: <sip:932222222@10.1.2.3;ftag=25433214-696;lr=on>
Contact: <sip:932222222@172.22.32.1:6000>
Content-Length: 0
*Jul 7 12:48:55.772: Received:
SIP/2.0 180 Ringing
To: <sip:932222222@10.1.2.3>;tag=c525a909
From: <sip:172.22.22.5>;tag=25433214-696
Via: SIP/2.0/UDP 172.22.22.5:5060;rport=52053
Call-ID: 4FCAC87E-EE1C11D9-808AE1DB-AA3EFB56(a)172.22.22.5
CSeq: 101 INVITE
Record-Route: <sip:932222222@10.1.2.3;ftag=25433214-696;lr=on>
Contact: <sip:932222222@172.22.32.1:6000>
Content-Length: 0
*Jul 7 12:49:02.368: Received:
SIP/2.0 200 OK
To: <sip:932222222@10.1.2.3>;tag=c525a909
From: <sip:172.22.22.5>;tag=25433214-696
Via: SIP/2.0/UDP 172.22.22.5:5060;rport=52053
Call-ID: 4FCAC87E-EE1C11D9-808AE1DB-AA3EFB56(a)172.22.22.5
CSeq: 101 INVITE
Record-Route: <sip:932222222@10.1.2.3;ftag=25433214-696;lr=on>
Contact: <sip:932222222@172.22.32.1:6000>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
INFO
Content-Type: application/sdp
Content-Length: 178
v=0
o=- 248133787 248133799 IN IP4 172.22.32.1
s=eyeBeam
c=IN IP4 172.22.32.1
t=0 0
m=audio 6002 RTP/AVP 18 0 8
a=alt:1 1 : 73A00FD8 FD138AA0 172.22.32.1 6002
a=sendrecv
*Jul 7 12:49:02.372: Sent:
ACK sip:932222222@10.1.2.3:5060;ftag=25433214-696;lr=on SIP/2.0
Via: SIP/2.0/UDP 172.22.22.5:5060
From: <sip:172.22.22.5>;tag=25433214-696
To: <sip:932222222@10.1.2.3>;tag=c525a909
Date: Thu, 07 Jul 2005 12:48:54 GMT
Call-ID: 4FCAC87E-EE1C11D9-808AE1DB-AA3EFB56(a)172.22.22.5
Route: <sip:932222222@172.22.32.1:6000>
Max-Forwards: 6
Content-Length: 0
CSeq: 101 ACK
From: "Ezequiel Colombo"
<ecolombo(a)telarsa.com.ar>
To: "Victor Huertas Garcia"
<vhuertas@hotmail.com>,<serusers@lists.iptel.org>
Subject: Re: [Serusers] URGENT! AS5350 <-> SER incompatibility problem
Date: Thu, 7 Jul 2005 09:18:59 -0300
Hola Victor
Can you send the trace ?
ngrep at SER or debug ccsip messages at Cisco, and show sip-ua status.
----- Original Message -----
From: "Victor Huertas Garcia" <vhuertas(a)hotmail.com>
To: <serusers(a)lists.iptel.org>
Sent: Thursday, July 07, 2005 8:05 AM
Subject: [Serusers] URGENT! AS5350 <-> SER incompatibility problem
|
|
| Hi everybody!
|
|
| recently I have been testing calls to/from ISDN with a AS5350. My SER
| version is 0.8.14 and when I make a call from ISDN towards a SIP UA
| registered in the SER the problem is that the AS5350 is no aware of Call
| Release.
|
| I explain it in more detail...
|
| 1. AS5350 sends an INVITE to the SER with destination number 93222222
| 2. SER forwards the INVITE to the corresponding SIP UA.
| 3. UA reponds with a 180 Ringing and it is forwarded to the AS5350 by the
| SER
| 4. When I pick up the phone, the SIP UA sends a 200 OK message, which is
| frowarded by the SER as well.
| 5. AS5350 replies with a ACK to the 200 OK but beware!! the AS5350 has
| changed UDP source port!! it is different from the one set in INVITE.
| 6. The SIP UA hangs up the call by sending a BYE message to the SER (it
is
| Record Routing)
| 7. The SER forwards the BYE to the AS5350 but with the old UDP port!!
| 8. AS5350 sends ICMP Destination Unreachable (Cause code: port
unreachable)
| every time that the SER retries the BYE.
|
|
| My guess is that the AS5350 expects the BYE packet through the new UDP
| port...
|
| Has anybody found such a problem?
|
| Thanks
|
| Victor
|
|
| _______________________________________________
| Serusers mailing list
| serusers(a)lists.iptel.org
| http://lists.iptel.org/mailman/listinfo/serusers
9:50 a.m.
Victor, i dont see where AS5350 change the port as you say. In this ACK Cisco
send also the 172.22.22.5:5060 in the Via HF.
*Jul 7 12:49:02.372: Sent:
ACK sip:932222222@10.1.2.3:5060;ftag=25433214-696;lr=on SIP/2.0
Via: SIP/2.0/UDP 172.22.22.5:5060
From: <sip:172.22.22.5>;tag=25433214-696
To: <sip:932222222@10.1.2.3>;tag=c525a909
Date: Thu, 07 Jul 2005 12:48:54 GMT
Call-ID: 4FCAC87E-EE1C11D9-808AE1DB-AA3EFB56(a)172.22.22.5
Route: <sip:932222222@172.22.32.1:6000>
Max-Forwards: 6
Content-Length: 0
CSeq: 101 ACK
You have some nat entity betwen AS5350 and SER ? I get confused with the rport
in the Via of 100 Trying ("Via: SIP/2.0/UDP 172.22.22.5:5060;rport=52053"). In
this test, the BYE is sent to AS5350 port 52053 ?
----- Original Message -----
From: "Victor Huertas Garcia" <vhuertas(a)hotmail.com>
To: <ecolombo(a)telarsa.com.ar>ar>; <serusers(a)lists.iptel.org>
Sent: Thursday, July 07, 2005 10:06 AM
Subject: Re: [Serusers] URGENT! AS5350 <-> SER incompatibility problem
| Here find the traces "debug ccsip messages"
|
|
|
| *Jul 7 12:48:54.936: Sent:
| INVITE sip:932222222@10.1.2.3:5060 SIP/2.0
|
| Via: SIP/2.0/UDP 172.22.22.5:5060
|
| From: <sip:172.22.22.5>;tag=25433214-696
|
| To: <sip:932222222@10.1.2.3>
|
| Date: Thu, 07 Jul 2005 12:48:54 GMT
|
| Call-ID: 4FCAC87E-EE1C11D9-808AE1DB-AA3EFB56(a)172.22.22.5
|
| Supported: timer,100rel
|
| Min-SE: 1800
|
| Cisco-Guid: 1338609710-3994816985-2156388827-2856254294
|
| User-Agent: Cisco-SIPGateway/IOS-12.x
|
| Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE,
| NOTIFY, INFO
|
| CSeq: 101 INVITE
|
| Max-Forwards: 6
|
| Remote-Party-ID: <sip:172.22.22.5>;party=calling;screen=no;privacy=off
|
| Timestamp: 1120740534
|
| Contact: <sip:172.22.22.5:5060>
|
| Expires: 180
|
| Allow-Events: telephone-event
|
| Content-Type: application/sdp
|
| Content-Length: 292
|
|
|
| v=0
|
| o=CiscoSystemsSIP-GW-UserAgent 362 6669 IN IP4 172.22.22.5
|
| s=SIP Call
|
| c=IN IP4 172.22.22.5
|
| t=0 0
|
| m=audio 17808 RTP/AVP 18 4 0 8
|
| c=IN IP4 172.22.22.5
|
| a=rtpmap:18 G729/8000
|
| a=fmtp:18 annexb=yes
|
| a=rtpmap:4 G723/8000
|
| a=fmtp:4 annexa=yes
|
| a=rtpmap:0 PCMU/8000
|
| a=rtpmap:8 PCMA/8000
|
|
| *Jul 7 12:48:54.936: Received:
| SIP/2.0 100 trying -- your call is important to us
|
| Via: SIP/2.0/UDP 172.22.22.5:5060;rport=52053
|
| From: <sip:172.22.22.5>;tag=25433214-696
|
| To: <sip:932222222@10.1.2.3>
|
| Call-ID: 4FCAC87E-EE1C11D9-808AE1DB-AA3EFB56(a)172.22.22.5
|
| CSeq: 101 INVITE
|
| Server: Sip EXpress router (0.8.14 (i386/linux))
|
| Content-Length: 0
|
| Warning: 392 10.1.2.3:5060 "Noisy feedback tells: pid=24175
| req_src_ip=172.22.22.5 req_src_port=52053 in_uri=sip:932222222@10.1.2.3:5060
| out_uri=sip:932222222@172.22.32.1:6000 via_cnt==1"
|
|
|
|
|
| *Jul 7 12:48:55.692: Received:
| SIP/2.0 180 Ringing
|
| To: <sip:932222222@10.1.2.3>;tag=c525a909
|
| From: <sip:172.22.22.5>;tag=25433214-696
|
| Via: SIP/2.0/UDP 172.22.22.5:5060;rport=52053
|
| Call-ID: 4FCAC87E-EE1C11D9-808AE1DB-AA3EFB56(a)172.22.22.5
|
| CSeq: 101 INVITE
|
| Record-Route: <sip:932222222@10.1.2.3;ftag=25433214-696;lr=on>
|
| Contact: <sip:932222222@172.22.32.1:6000>
|
| Content-Length: 0
|
|
|
|
|
| *Jul 7 12:48:55.772: Received:
| SIP/2.0 180 Ringing
|
| To: <sip:932222222@10.1.2.3>;tag=c525a909
|
| From: <sip:172.22.22.5>;tag=25433214-696
|
| Via: SIP/2.0/UDP 172.22.22.5:5060;rport=52053
|
| Call-ID: 4FCAC87E-EE1C11D9-808AE1DB-AA3EFB56(a)172.22.22.5
|
| CSeq: 101 INVITE
|
| Record-Route: <sip:932222222@10.1.2.3;ftag=25433214-696;lr=on>
|
| Contact: <sip:932222222@172.22.32.1:6000>
|
| Content-Length: 0
|
|
|
|
|
| *Jul 7 12:49:02.368: Received:
| SIP/2.0 200 OK
|
| To: <sip:932222222@10.1.2.3>;tag=c525a909
|
| From: <sip:172.22.22.5>;tag=25433214-696
|
| Via: SIP/2.0/UDP 172.22.22.5:5060;rport=52053
|
| Call-ID: 4FCAC87E-EE1C11D9-808AE1DB-AA3EFB56(a)172.22.22.5
|
| CSeq: 101 INVITE
|
| Record-Route: <sip:932222222@10.1.2.3;ftag=25433214-696;lr=on>
|
| Contact: <sip:932222222@172.22.32.1:6000>
|
| Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
| INFO
|
| Content-Type: application/sdp
|
| Content-Length: 178
|
|
|
| v=0
|
| o=- 248133787 248133799 IN IP4 172.22.32.1
|
| s=eyeBeam
|
| c=IN IP4 172.22.32.1
|
| t=0 0
|
| m=audio 6002 RTP/AVP 18 0 8
|
| a=alt:1 1 : 73A00FD8 FD138AA0 172.22.32.1 6002
|
| a=sendrecv
|
|
| *Jul 7 12:49:02.372: Sent:
| ACK sip:932222222@10.1.2.3:5060;ftag=25433214-696;lr=on SIP/2.0
|
| Via: SIP/2.0/UDP 172.22.22.5:5060
|
| From: <sip:172.22.22.5>;tag=25433214-696
|
| To: <sip:932222222@10.1.2.3>;tag=c525a909
|
| Date: Thu, 07 Jul 2005 12:48:54 GMT
|
| Call-ID: 4FCAC87E-EE1C11D9-808AE1DB-AA3EFB56(a)172.22.22.5
|
| Route: <sip:932222222@172.22.32.1:6000>
|
| Max-Forwards: 6
|
| Content-Length: 0
|
| CSeq: 101 ACK
|
|
|
|
|
| >From: "Ezequiel Colombo" <ecolombo(a)telarsa.com.ar>
| >To: "Victor Huertas Garcia"
<vhuertas@hotmail.com>,<serusers@lists.iptel.org>
| >Subject: Re: [Serusers] URGENT! AS5350 <-> SER incompatibility problem
| >Date: Thu, 7 Jul 2005 09:18:59 -0300
| >
| >Hola Victor
| >Can you send the trace ?
| >ngrep at SER or debug ccsip messages at Cisco, and show sip-ua status.
| >
| >----- Original Message -----
| >From: "Victor Huertas Garcia" <vhuertas(a)hotmail.com>
| >To: <serusers(a)lists.iptel.org>
| >Sent: Thursday, July 07, 2005 8:05 AM
| >Subject: [Serusers] URGENT! AS5350 <-> SER incompatibility problem
| >
| >
| >|
| >|
| >| Hi everybody!
| >|
| >|
| >| recently I have been testing calls to/from ISDN with a AS5350. My SER
| >| version is 0.8.14 and when I make a call from ISDN towards a SIP UA
| >| registered in the SER the problem is that the AS5350 is no aware of Call
| >| Release.
| >|
| >| I explain it in more detail...
| >|
| >| 1. AS5350 sends an INVITE to the SER with destination number 93222222
| >| 2. SER forwards the INVITE to the corresponding SIP UA.
| >| 3. UA reponds with a 180 Ringing and it is forwarded to the AS5350 by the
| >| SER
| >| 4. When I pick up the phone, the SIP UA sends a 200 OK message, which is
| >| frowarded by the SER as well.
| >| 5. AS5350 replies with a ACK to the 200 OK but beware!! the AS5350 has
| >| changed UDP source port!! it is different from the one set in INVITE.
| >| 6. The SIP UA hangs up the call by sending a BYE message to the SER (it
| >is
| >| Record Routing)
| >| 7. The SER forwards the BYE to the AS5350 but with the old UDP port!!
| >| 8. AS5350 sends ICMP Destination Unreachable (Cause code: port
| >unreachable)
| >| every time that the SER retries the BYE.
| >|
| >|
| >| My guess is that the AS5350 expects the BYE packet through the new UDP
| >| port...
| >|
| >| Has anybody found such a problem?
| >|
| >| Thanks
| >|
| >| Victor
| >|
| >|
| >| _______________________________________________
| >| Serusers mailing list
| >| serusers(a)lists.iptel.org
| >| http://lists.iptel.org/mailman/listinfo/serusers
| >
|
|
6979
days inactive
6979
days old
3 comments
2 participants
participants (2)
-
Ezequiel Colombo -
Victor Huertas Garcia