2 Sep
2004
2 Sep
'04
6:49 p.m.
HI,
I have an interconexion with a proxy-server registrar IPTEL to a Cisco
GATEWAY AS5300 for the PSTN.
When I make a call by a Cisco IP phone, I have no problem but when I make a
call with a call-box(Same as Cisco ATA) or a softphone Xpro (XTEN), I can
speak 20 s and after the call is cut.
I have analysed this problem, the problem is after the SIP ringing frame,
the softphone send a SIP frame ACK with a bad request-line.
IP ADDRESS 80.118.128.1 : CISCO AS5300 GATEWAY
IP ADDRESS 80.118.128.5 : SERVER IPTEL
The RED parameter in the first DEBUG is BAD, he as to be as the same of the
second debug.
That's my debug of the SOFT PHONE: BAD ACKNOWLEDGE
#########################################################
U 80.118.128.1:5060 -> 80.118.128.5:5060
SIP/2.0 200 OK..Via: SIP/2.0/UDP
80.118.128.5;branch=z9hG4bKcce9.7a6be6f3.0,SIP/2.0/UDP
62.39.69.2:21791;rport=21791;branch=z9hG4bKB0A9B5A4D70A4FF8A745CCDD512FED41.
.Fr
om: Nicolas RUIZ <sip:0170708661@sip.vivaction.net>;tag=1756293558..To:
<sip:0156384185@sip.vivaction.net>;tag=E855BD4-163A..Date: Thu, 02 Sep 2004
12:39:19 GMT..Call-
ID: 475E7EC7-5CA5-457D-8F0D-BCF6B8FA6209@192.168.18.21..Server:
Cisco-SIPGateway/IOS-12.x..CSeq: 24531 INVITE..Allow: INVITE, OPTIONS, BYE,
CANCEL, ACK, PRACK, COMET,
REFER, SUBSCRIBE, NOTIFY, INFO..Allow-Events: telephone-event..Contact:
<sip:0156384185@80.118.128.1:5060>..Record-Route:
<sip:0156384185@80.118.128.5;ftag=1756293558;
lr=on>..Content-Type: application/sdp..Content-Length:
256....v=0..o=CiscoSystemsSIP-GW-UserAgent 16 8937 IN IP4
80.118.128.1..s=SIP Call..c=IN IP4 80.118.128.1..t=0 0
..m=audio 18194 RTP/AVP 18 101..c=IN IP4 80.118.128.1..a=rtpmap:18
G729/8000..a=fmtp:18 annexb=no..a=rtpmap:101
telephone-event/8000..a=fmtp:101 0-15..
#
U 80.118.128.5:5060 -> 62.39.69.2:21791
SIP/2.0 200 OK..Via: SIP/2.0/UDP
62.39.69.2:21791;rport=21791;branch=z9hG4bKB0A9B5A4D70A4FF8A745CCDD512FED41.
.From: Nicolas RUIZ <sip:0170708661@sip.vivaction.net>;tag
=1756293558..To:
<sip:0156384185@sip.vivaction.net>;tag=E855BD4-163A..Date: Thu, 02 Sep 2004
12:39:19 GMT..Call-ID: 475E7EC7-5CA5-457D-8F0D-BCF6B8FA6209(a)192.168.18.21.
.Server: Cisco-SIPGateway/IOS-12.x..CSeq: 24531 INVITE..Allow: INVITE,
OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY,
INFO..Allow-Events: telephone
-event..Contact: <sip:0156384185@80.118.128.1:5060>..Record-Route:
<sip:0156384185@80.118.128.5;ftag=1756293558;lr=on>..Content-Type:
application/sdp..Content-Length:
256....v=0..o=CiscoSystemsSIP-GW-UserAgent 16 8937 IN IP4
80.118.128.1..s=SIP Call..c=IN IP4 80.118.128.1..t=0 0..m=audio 18194
RTP/AVP 18 101..c=IN IP4 80.118.128.1..
a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:101
telephone-event/8000..a=fmtp:101 0-15..
####
U 62.39.69.2:21791 -> 80.118.128.5:5060
ACK sip:0156384185@80.118.128.1:5060 SIP/2.0..Via: SIP/2.0/UDP
62.39.69.2:21791;rport;branch=z9hG4bKDDE6A61442CF4B3985D47F7A633E9A9F..From:
Nicolas RUIZ <sip:017070866
1@sip.vivaction.net>;tag=1756293558..To:
<sip:0156384185@sip.vivaction.net>;tag=E855BD4-163A..Contact:
<sip:0170708661@62.39.69.2:21791>..Route: <sip:0156384185@80.118
.128.5;ftag=1756293558;lr=on>..Call-ID:
475E7EC7-5CA5-457D-8F0D-BCF6B8FA6209@192.168.18.21..CSeq: 24531
ACK..Max-Forwards: 70..Content-Length: 0....
######
That's my debug of the IP PHONE: GOOD ACKNOWLEDGE
U 80.118.128.1:5060 -> 80.118.128.5:5060
SIP/2.0 200 OK..Via: SIP/2.0/UDP
80.118.128.5;branch=z9hG4bK119a.2eb3c633.0,SIP/2.0/UDP
62.39.70.245:20105;branch=z9hG4bK21610a53..From: "ip.phone"
<sip:0170708662@sip
.vivaction.net>;tag=000f8f58915f002e0e896df1-21abefc7..To:
<sip:0156384185@sip.vivaction.net>;tag=EB213D4-226C..Date: Thu, 02 Sep 2004
13:28:10 GMT..Call-ID: 000f8f58-
915f0025-7e34225d-1cdf9dae@192.168.254.12..Server:
Cisco-SIPGateway/IOS-12.x..CSeq: 102 INVITE..Allow: INVITE, OPTIONS, BYE,
CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIB
E, NOTIFY, INFO..Allow-Events: telephone-event..Contact:
<sip:0156384185@80.118.128.1:5060>..Record-Route:
<sip:0156384185@80.118.128.5;ftag=000f8f58915f002e0e896df1-2
1abefc7;lr=on>..Content-Type: application/sdp..Content-Length:
258....v=0..o=CiscoSystemsSIP-GW-UserAgent 1735 3645 IN IP4
80.118.128.1..s=SIP Call..c=IN IP4 80.118.12
8.1..t=0 0..m=audio 18760 RTP/AVP 18 101..c=IN IP4
80.118.128.1..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:101
telephone-event/8000..a=fmtp:101 0-15..
#
U 80.118.128.5:5060 -> 62.39.70.245:20105
SIP/2.0 200 OK..Via: SIP/2.0/UDP
62.39.70.245:20105;branch=z9hG4bK21610a53..From: "ip.phone"
<sip:0170708662@sip.vivaction.net>;tag=000f8f58915f002e0e896df1-21abefc7..
To: <sip:0156384185@sip.vivaction.net>;tag=EB213D4-226C..Date: Thu, 02 Sep
2004 13:28:10 GMT..Call-ID:
000f8f58-915f0025-7e34225d-1cdf9dae@192.168.254.12..Server: Cisc
o-SIPGateway/IOS-12.x..CSeq: 102 INVITE..Allow: INVITE, OPTIONS, BYE,
CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO..Allow-Events:
telephone-event..Contact
: <sip:0156384185@80.118.128.1:5060>..Record-Route:
<sip:0156384185@80.118.128.5;ftag=000f8f58915f002e0e896df1-21abefc7;lr=on>..
Content-Type: application/sdp..Content-
Length: 258....v=0..o=CiscoSystemsSIP-GW-UserAgent 1735 3645 IN IP4
80.118.128.1..s=SIP Call..c=IN IP4 80.118.128.1..t=0 0..m=audio 18760
RTP/AVP 18 101..c=IN IP4 80.1
18.128.1..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:101
telephone-event/8000..a=fmtp:101 0-15..
####
U 62.39.70.245:20600 -> 80.118.128.5:5060
ACK
sip:0156384185@80.118.128.5:5060;ftag=000f8f58915f002e0e896df1-21abefc7;lr=o
n SIP/2.0..Via: SIP/2.0/UDP 62.39.70.245:20105;branch=z9hG4bK47003517..From:
"ip.phone"
<sip:0170708662@sip.vivaction.net>;tag=000f8f58915f002e0e896df1-21abefc7..To
: <sip:0156384185@sip.vivaction.net>;tag=EB213D4-226C..Call-ID:
000f8f58-915f0025-7e34225d
-1cdf9dae@192.168.254.12..CSeq: 102 ACK..User-Agent: CSCO/6..Route:
<sip:0156384185@80.118.128.1:5060>..Content-Length: 0....
#
U 80.118.128.5:5060 -> 80.118.128.1:5060
ACK sip:0156384185@80.118.128.1:5060 SIP/2.0..Max-Forwards:
10..Record-Route:
<sip:0156384185@80.118.128.5;ftag=000f8f58915f002e0e896df1-21abefc7;lr=on>..
Via: SIP/2.0/
UDP 80.118.128.5;branch=0..Via: SIP/2.0/UDP
62.39.70.245:20105;branch=z9hG4bK47003517..From: "ip.phone"
<sip:0170708662@sip.vivaction.net>;tag=000f8f58915f002e0e896df1
-21abefc7..To:
<sip:0156384185@sip.vivaction.net>;tag=EB213D4-226C..Call-ID:
000f8f58-915f0025-7e34225d-1cdf9dae@192.168.254.12..CSeq: 102
ACK..User-Agent: CSCO/6..Con
tent-Length: 0....
Thans for your help
Nicolas RUIZ
PARIS, FRANCE
2 Sep
2 Sep
8:09 p.m.
Both of them are correct. In the first ACK the user agent correctly
recognized that the next proxy server is loose router and put the
Contact from 200 OK into the Request-URI of ACK.
In the other example, the user agent did not recognized that the proxy
is loose router (or does not implement loose routing) and behaves a
specificied in RFC2543, it puts the content of the Route header field
into the Request-URI and the Contact from 200 OK is put into another
Route header field.
Both cases should work.
Jan.
On 02-09 17:49, Nicolas RUIZ wrote:
HI,
I have an interconexion with a proxy-server registrar IPTEL to a Cisco
GATEWAY AS5300 for the PSTN.
When I make a call by a Cisco IP phone, I have no problem but when I make a
call with a call-box(Same as Cisco ATA) or a softphone Xpro (XTEN), I can
speak 20 s and after the call is cut.
I have analysed this problem, the problem is after the SIP ringing frame,
the softphone send a SIP frame ACK with a bad request-line.
IP ADDRESS 80.118.128.1 : CISCO AS5300 GATEWAY
IP ADDRESS 80.118.128.5 : SERVER IPTEL
The RED parameter in the first DEBUG is BAD, he as to be as the same of the
second debug.
That's my debug of the SOFT PHONE: BAD ACKNOWLEDGE
#########################################################
U 80.118.128.1:5060 -> 80.118.128.5:5060
SIP/2.0 200 OK..Via: SIP/2.0/UDP
80.118.128.5;branch=z9hG4bKcce9.7a6be6f3.0,SIP/2.0/UDP
62.39.69.2:21791;rport=21791;branch=z9hG4bKB0A9B5A4D70A4FF8A745CCDD512FED41.
.Fr
om: Nicolas RUIZ <sip:0170708661@sip.vivaction.net>;tag=1756293558..To:
<sip:0156384185@sip.vivaction.net>;tag=E855BD4-163A..Date: Thu, 02 Sep 2004
12:39:19 GMT..Call-
ID: 475E7EC7-5CA5-457D-8F0D-BCF6B8FA6209@192.168.18.21..Server:
Cisco-SIPGateway/IOS-12.x..CSeq: 24531 INVITE..Allow: INVITE, OPTIONS, BYE,
CANCEL, ACK, PRACK, COMET,
REFER, SUBSCRIBE, NOTIFY, INFO..Allow-Events: telephone-event..Contact:
<sip:0156384185@80.118.128.1:5060>..Record-Route:
<sip:0156384185@80.118.128.5;ftag=1756293558;
lr=on>..Content-Type: application/sdp..Content-Length:
256....v=0..o=CiscoSystemsSIP-GW-UserAgent 16 8937 IN IP4
80.118.128.1..s=SIP Call..c=IN IP4 80.118.128.1..t=0 0
..m=audio 18194 RTP/AVP 18 101..c=IN IP4 80.118.128.1..a=rtpmap:18
G729/8000..a=fmtp:18 annexb=no..a=rtpmap:101
telephone-event/8000..a=fmtp:101 0-15..
#
U 80.118.128.5:5060 -> 62.39.69.2:21791
SIP/2.0 200 OK..Via: SIP/2.0/UDP
62.39.69.2:21791;rport=21791;branch=z9hG4bKB0A9B5A4D70A4FF8A745CCDD512FED41.
.From: Nicolas RUIZ <sip:0170708661@sip.vivaction.net>;tag
=1756293558..To:
<sip:0156384185@sip.vivaction.net>;tag=E855BD4-163A..Date: Thu, 02 Sep 2004
12:39:19 GMT..Call-ID: 475E7EC7-5CA5-457D-8F0D-BCF6B8FA6209(a)192.168.18.21.
.Server: Cisco-SIPGateway/IOS-12.x..CSeq: 24531 INVITE..Allow: INVITE,
OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY,
INFO..Allow-Events: telephone
-event..Contact: <sip:0156384185@80.118.128.1:5060>..Record-Route:
<sip:0156384185@80.118.128.5;ftag=1756293558;lr=on>..Content-Type:
application/sdp..Content-Length:
256....v=0..o=CiscoSystemsSIP-GW-UserAgent 16 8937 IN IP4
80.118.128.1..s=SIP Call..c=IN IP4 80.118.128.1..t=0 0..m=audio 18194
RTP/AVP 18 101..c=IN IP4 80.118.128.1..
a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:101
telephone-event/8000..a=fmtp:101 0-15..
####
U 62.39.69.2:21791 -> 80.118.128.5:5060
ACK sip:0156384185@80.118.128.1:5060 SIP/2.0..Via: SIP/2.0/UDP
62.39.69.2:21791;rport;branch=z9hG4bKDDE6A61442CF4B3985D47F7A633E9A9F..From:
Nicolas RUIZ <sip:017070866
1@sip.vivaction.net>;tag=1756293558..To:
<sip:0156384185@sip.vivaction.net>;tag=E855BD4-163A..Contact:
<sip:0170708661@62.39.69.2:21791>..Route: <sip:0156384185@80.118
.128.5;ftag=1756293558;lr=on>..Call-ID:
475E7EC7-5CA5-457D-8F0D-BCF6B8FA6209@192.168.18.21..CSeq: 24531
ACK..Max-Forwards: 70..Content-Length: 0....
######
That's my debug of the IP PHONE: GOOD ACKNOWLEDGE
U 80.118.128.1:5060 -> 80.118.128.5:5060
SIP/2.0 200 OK..Via: SIP/2.0/UDP
80.118.128.5;branch=z9hG4bK119a.2eb3c633.0,SIP/2.0/UDP
62.39.70.245:20105;branch=z9hG4bK21610a53..From: "ip.phone"
<sip:0170708662@sip
.vivaction.net>;tag=000f8f58915f002e0e896df1-21abefc7..To:
<sip:0156384185@sip.vivaction.net>;tag=EB213D4-226C..Date: Thu, 02 Sep 2004
13:28:10 GMT..Call-ID: 000f8f58-
915f0025-7e34225d-1cdf9dae@192.168.254.12..Server:
Cisco-SIPGateway/IOS-12.x..CSeq: 102 INVITE..Allow: INVITE, OPTIONS, BYE,
CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIB
E, NOTIFY, INFO..Allow-Events: telephone-event..Contact:
<sip:0156384185@80.118.128.1:5060>..Record-Route:
<sip:0156384185@80.118.128.5;ftag=000f8f58915f002e0e896df1-2
1abefc7;lr=on>..Content-Type: application/sdp..Content-Length:
258....v=0..o=CiscoSystemsSIP-GW-UserAgent 1735 3645 IN IP4
80.118.128.1..s=SIP Call..c=IN IP4 80.118.12
8.1..t=0 0..m=audio 18760 RTP/AVP 18 101..c=IN IP4
80.118.128.1..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:101
telephone-event/8000..a=fmtp:101 0-15..
#
U 80.118.128.5:5060 -> 62.39.70.245:20105
SIP/2.0 200 OK..Via: SIP/2.0/UDP
62.39.70.245:20105;branch=z9hG4bK21610a53..From: "ip.phone"
<sip:0170708662@sip.vivaction.net>;tag=000f8f58915f002e0e896df1-21abefc7..
To: <sip:0156384185@sip.vivaction.net>;tag=EB213D4-226C..Date: Thu, 02 Sep
2004 13:28:10 GMT..Call-ID:
000f8f58-915f0025-7e34225d-1cdf9dae@192.168.254.12..Server: Cisc
o-SIPGateway/IOS-12.x..CSeq: 102 INVITE..Allow: INVITE, OPTIONS, BYE,
CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO..Allow-Events:
telephone-event..Contact
: <sip:0156384185@80.118.128.1:5060>..Record-Route:
<sip:0156384185@80.118.128.5;ftag=000f8f58915f002e0e896df1-21abefc7;lr=on>..
Content-Type: application/sdp..Content-
Length: 258....v=0..o=CiscoSystemsSIP-GW-UserAgent 1735 3645 IN IP4
80.118.128.1..s=SIP Call..c=IN IP4 80.118.128.1..t=0 0..m=audio 18760
RTP/AVP 18 101..c=IN IP4 80.1
18.128.1..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:101
telephone-event/8000..a=fmtp:101 0-15..
####
U 62.39.70.245:20600 -> 80.118.128.5:5060
ACK
sip:0156384185@80.118.128.5:5060;ftag=000f8f58915f002e0e896df1-21abefc7;lr=o
n SIP/2.0..Via: SIP/2.0/UDP 62.39.70.245:20105;branch=z9hG4bK47003517..From:
"ip.phone"
<sip:0170708662@sip.vivaction.net>;tag=000f8f58915f002e0e896df1-21abefc7..To
: <sip:0156384185@sip.vivaction.net>;tag=EB213D4-226C..Call-ID:
000f8f58-915f0025-7e34225d
-1cdf9dae@192.168.254.12..CSeq: 102 ACK..User-Agent: CSCO/6..Route:
<sip:0156384185@80.118.128.1:5060>..Content-Length: 0....
#
U 80.118.128.5:5060 -> 80.118.128.1:5060
ACK sip:0156384185@80.118.128.1:5060 SIP/2.0..Max-Forwards:
10..Record-Route:
<sip:0156384185@80.118.128.5;ftag=000f8f58915f002e0e896df1-21abefc7;lr=on>..
Via: SIP/2.0/
UDP 80.118.128.5;branch=0..Via: SIP/2.0/UDP
62.39.70.245:20105;branch=z9hG4bK47003517..From: "ip.phone"
<sip:0170708662@sip.vivaction.net>;tag=000f8f58915f002e0e896df1
-21abefc7..To:
<sip:0156384185@sip.vivaction.net>;tag=EB213D4-226C..Call-ID:
000f8f58-915f0025-7e34225d-1cdf9dae@192.168.254.12..CSeq: 102
ACK..User-Agent: CSCO/6..Con
tent-Length: 0....
Thans for your help
Nicolas RUIZ
PARIS, FRANCE
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Nicolas RUIZ