17 Nov
2004
17 Nov
'04
11:45 p.m.
Hi All.
I'm using dev12 and I'm interfacing with a company that uses Sonus equipment to
terminate to the
PSTN. This company is telling me that SER is not performing to RFC3261 because ACK
messages are
not including all the record-route information as stated in section 12.1.2.
Following is an email I recieved from their network engineers. Attached is also a
complete
conversation between my SER proxy and their Sonus box. The actual problem is that Sonus
disconnects the call after a few seconds because of an ACK routing issue.
Anyhow can someone advise me on weather or not this is truly a SER issue or a Sonus
issue?
Regards,
Paul
+++++++++EMAIL FROM THE GUYS USING SONUS++++++++++++++
No, this is definitely not a Sonus issue.
The response sent from the Sonus to the SER contains the correct set of
Record-Route (every hop that the original INVITE has traversed). When the
SER sends back the ACK, the SER if it acts as a UAC, then must include this
set of Record-Route as Route in the message per RFC3261 section 12.1.2.
I believe you will have problem with this if you interface the SER with any
SIP proxy. I assume that it works with Broadvox because you are connected
directly to their gateway and no other proxy is in between. If you were to
connect directly with the Sonus, then it would work.
The problem should be fixed on your end. You just need to look into the
code where it generates the ACK and follow RFC3261 section 12.1.2 on how to
build and send it.
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U 2004/11/17 14:59:50.990562 68.80.200.100:1063 -> 68.80.201.101:5060
INVITE sip:14075551212@sip.mycompany.com;user=phone SIP/2.0.
Via: SIP/2.0/UDP 192.168.0.83;branch=z9hG4bKedd18d2af50b4ade.
From: "Paul (1002)"
<sip:9990010001@sip.mycompany.com;user=phone>;tag=4edd147cdac0025f.
To: <sip:14075551212@sip.mycompany.com;user=phone>.
Contact: <sip:9990010001@192.168.0.83;user=phone>.
Call-ID: aa31202374f54793(a)192.168.0.83.
CSeq: 12009 INVITE.
User-Agent: Grandstream BT100 1.0.5.11.
Max-Forwards: 70.
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE.
Content-Type: application/sdp.
Content-Length: 377.
.
v=0.
o=9990010001 8000 8000 IN IP4 192.168.0.83.
s=SIP Call.
c=IN IP4 192.168.0.83.
t=0 0.
m=audio 5004 RTP/AVP 98 18 4 15 2 8 9 101.
a=rtpmap:98 iLBC/8000.
a=fmtp:98 mode=20.
a=rtpmap:18 G729/8000.
a=rtpmap:4 G723/8000.
a=rtpmap:15 G728/8000.
a=rtpmap:2 G726-32/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:9 G722/8000.
a=ptime:40.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-11.
#
U 2004/11/17 14:59:50.991323 68.80.201.101:5060 -> 68.80.200.100:1063
SIP/2.0 407 Proxy Authentication Required.
Via: SIP/2.0/UDP
192.168.0.83;branch=z9hG4bKedd18d2af50b4ade;rport=1063;received=68.80.200.100.
From: "Paul (1002)"
<sip:9990010001@sip.mycompany.com;user=phone>;tag=4edd147cdac0025f.
To:
<sip:14075551212@sip.mycompany.com;user=phone>;tag=bf952ed189d8425c881b09485aa0b6f1.3e61.
Call-ID: aa31202374f54793(a)192.168.0.83.
CSeq: 12009 INVITE.
Proxy-Authenticate: Digest realm="sip.mycompany.com",
nonce="419baee2139477c22db6d0ec9a47b23003512431".
Server: mycompany SIP Router (0.8.99-dev12 (i386/linux)).
Content-Length: 0.
Warning: 392 68.80.201.101:5060 "Noisy feedback tells: pid=8495
req_src_ip=68.80.200.100 req_src_port=1063
in_uri=sip:14075551212@sip.mycompany.com;user=phone
out_uri=sip:14075551212@sip.mycompany.com;user=phone via_cnt==1".
.
#
U 2004/11/17 14:59:50.997782 68.80.200.100:1063 -> 68.80.201.101:5060
ACK sip:14075551212@sip.mycompany.com;user=phone SIP/2.0.
Via: SIP/2.0/UDP 192.168.0.83;branch=z9hG4bKedd18d2af50b4ade.
From: "Paul (1002)"
<sip:9990010001@sip.mycompany.com;user=phone>;tag=4edd147cdac0025f.
To:
<sip:14075551212@sip.mycompany.com;user=phone>;tag=bf952ed189d8425c881b09485aa0b6f1.3e61.
Contact: <sip:9990010001@192.168.0.83;user=phone>.
Call-ID: aa31202374f54793(a)192.168.0.83.
CSeq: 12009 ACK.
User-Agent: Grandstream BT100 1.0.5.11.
Max-Forwards: 70.
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE.
Content-Length: 0.
.
#
U 2004/11/17 14:59:51.001721 68.80.200.100:1063 -> 68.80.201.101:5060
INVITE sip:14075551212@sip.mycompany.com;user=phone SIP/2.0.
Via: SIP/2.0/UDP 192.168.0.83;branch=z9hG4bK20ba9258b87f741d.
From: "Paul (1002)"
<sip:9990010001@sip.mycompany.com;user=phone>;tag=4edd147cdac0025f.
To: <sip:14075551212@sip.mycompany.com;user=phone>.
Contact: <sip:9990010001@192.168.0.83;user=phone>.
Proxy-Authorization: DIGEST username="9990010001",
realm="sip.mycompany.com", algorithm=MD5,
uri="sip:14075551212@sip.mycompany.com;user=phone",
nonce="419baee2139477c22db6d0ec9a47b23003512431",
response="239c34ca795c25a403283072aceb469b".
Call-ID: aa31202374f54793(a)192.168.0.83.
CSeq: 12010 INVITE.
User-Agent: Grandstream BT100 1.0.5.11.
Max-Forwards: 70.
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE.
Content-Type: application/sdp.
Content-Length: 377.
.
v=0.
o=9990010001 8000 8000 IN IP4 192.168.0.83.
s=SIP Call.
c=IN IP4 192.168.0.83.
t=0 0.
m=audio 5004 RTP/AVP 98 18 4 15 2 8 9 101.
a=rtpmap:98 iLBC/8000.
a=fmtp:98 mode=20.
a=rtpmap:18 G729/8000.
a=rtpmap:4 G723/8000.
a=rtpmap:15 G728/8000.
a=rtpmap:2 G726-32/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:9 G722/8000.
a=ptime:40.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-11.
#
U 2004/11/17 14:59:51.003660 68.80.201.101:5060 -> 68.80.200.100:1063
SIP/2.0 100 trying -- your call is important to us.
Via: SIP/2.0/UDP
192.168.0.83;branch=z9hG4bK20ba9258b87f741d;rport=1063;received=68.80.200.100.
From: "Paul (1002)"
<sip:9990010001@sip.mycompany.com;user=phone>;tag=4edd147cdac0025f.
To: <sip:14075551212@sip.mycompany.com;user=phone>.
Call-ID: aa31202374f54793(a)192.168.0.83.
CSeq: 12010 INVITE.
Server: mycompany SIP Router (0.8.99-dev12 (i386/linux)).
Content-Length: 0.
Warning: 392 68.80.201.101:5060 "Noisy feedback tells: pid=8493
req_src_ip=68.80.200.100 req_src_port=1063
in_uri=sip:14075551212@sip.mycompany.com;user=phone
out_uri=sip:4075551212@216.229.127.60;user=phone via_cnt==1".
.
#
U 2004/11/17 14:59:51.003755 68.80.201.101:5060 -> 216.229.127.60:5060
INVITE sip:4075551212@216.229.127.60;user=phone SIP/2.0.
Record-Route: <sip:68.80.201.101;ftag=4edd147cdac0025f;lr=on>.
Via: SIP/2.0/UDP 68.80.201.101;branch=z9hG4bK6628.1a8491a7.0.
Via: SIP/2.0/UDP
192.168.0.83;rport=1063;received=68.80.200.100;branch=z9hG4bK20ba9258b87f741d.
From: "Paul (1002)"
<sip:9990010001@sip.mycompany.com;user=phone>;tag=4edd147cdac0025f.
To: <sip:14075551212@sip.mycompany.com;user=phone>.
Contact: <sip:9990010001@68.80.200.100:1063;user=phone>.
Proxy-Authorization: DIGEST username="9990010001",
realm="sip.mycompany.com", algorithm=MD5,
uri="sip:14075551212@sip.mycompany.com;user=phone",
nonce="419baee2139477c22db6d0ec9a47b23003512431",
response="239c34ca795c25a403283072aceb469b".
Call-ID: aa31202374f54793(a)192.168.0.83.
CSeq: 12010 INVITE.
User-Agent: Grandstream BT100 1.0.5.11.
Max-Forwards: 16.
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE.
Content-Type: application/sdp.
Content-Length: 417.
.
v=0.
o=9990010001 8000 8000 IN IP4 192.168.0.83.
s=SIP Call.
c=IN IP4 68.80.201.101.
t=0 0.
m=audio 35228 RTP/AVP 98 18 4 15 2 8 9 101.
a=rtpmap:98 iLBC/8000.
a=fmtp:98 mode=20.
a=rtpmap:18 G729/8000.
a=rtpmap:4 G723/8000.
a=rtpmap:15 G728/8000.
a=rtpmap:2 G726-32/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:9 G722/8000.
a=ptime:40.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-11.
a=direction:active.
a=nortpproxy:yes.
#
U 2004/11/17 14:59:51.052928 216.229.127.60:5060 -> 68.80.201.101:5060
SIP/2.0 100 Try.
Via: SIP/2.0/UDP 68.80.201.101;branch=z9hG4bK6628.1a8491a7.0.
Via: SIP/2.0/UDP
192.168.0.83;rport=1063;received=68.80.200.100;branch=z9hG4bK20ba9258b87f741d.
To: <sip:14075551212@sip.mycompany.com;user=phone>.
From: "Paul (1002)"
<sip:9990010001@sip.mycompany.com;user=phone>;tag=4edd147cdac0025f.
Call-ID: aa31202374f54793(a)192.168.0.83.
CSeq: 12010 INVITE.
Proxy-Authorization: Digest username="9990010001",
realm="sip.mycompany.com", algorithm=MD5,
uri="sip:14075551212@sip.mycompany.com;user=phone",
nonce="419baee2139477c22db6d0ec9a47b23003512431",
response="239c34ca795c25a403283072aceb469b".
User-Agent: Grandstream BT100 1.0.5.11 .
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE.
Content-Type: application/sdp.
Content-Length: 0.
.
#####
U 2004/11/17 14:59:54.187710 216.229.127.60:5060 -> 68.80.201.101:5060
SIP/2.0 183 Session Progress.
Via: SIP/2.0/UDP 68.80.201.101;branch=z9hG4bK6628.1a8491a7.0.
Via: SIP/2.0/UDP
192.168.0.83;rport=1063;received=68.80.200.100;branch=z9hG4bK20ba9258b87f741d.
To: <sip:14075551212@sip.mycompany.com;user=phone>;tag=0bf5212d.
From: "Paul (1002)"
<sip:9990010001@sip.mycompany.com;user=phone>;tag=4edd147cdac0025f.
Call-ID: aa31202374f54793(a)192.168.0.83.
CSeq: 12010 INVITE.
Contact: sip:4075551212@216.229.50.90:4060.
Record-Route: <sip:216.229.127.60:5060;lr>.
Record-Route: <sip:68.80.201.101;ftag=4edd147cdac0025f;lr=on>.
Allow: OPTIONS, INVITE, CANCEL, ACK, BYE, PRACK, INFO.
Content-Disposition: session;handling=required.
Content-Type: application/sdp.
Content-Length: 225.
.
v=0.
o=Sonus_UAC 57 25508 IN IP4 216.229.50.90.
s=SIP Media Capabilities.
c=IN IP4 216.229.118.100.
t=0 0.
m=audio 18212 RTP/AVP 18 101.
a=rtpmap:18 G729/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sendrecv.
#
U 2004/11/17 14:59:54.188328 68.80.201.101:5060 -> 68.80.200.100:1063
SIP/2.0 183 Session Progress.
Via: SIP/2.0/UDP
192.168.0.83;rport=1063;received=68.80.200.100;branch=z9hG4bK20ba9258b87f741d.
To: <sip:14075551212@sip.mycompany.com;user=phone>;tag=0bf5212d.
From: "Paul (1002)"
<sip:9990010001@sip.mycompany.com;user=phone>;tag=4edd147cdac0025f.
Call-ID: aa31202374f54793(a)192.168.0.83.
CSeq: 12010 INVITE.
Contact: sip:4075551212@216.229.127.60:5060.
Record-Route: <sip:216.229.127.60:5060;lr>.
Record-Route: <sip:68.80.201.101;ftag=4edd147cdac0025f;lr=on>.
Allow: OPTIONS, INVITE, CANCEL, ACK, BYE, PRACK, INFO.
Content-Disposition: session;handling=required.
Content-Type: application/sdp.
Content-Length: 241.
.
v=0.
o=Sonus_UAC 57 25508 IN IP4 216.229.50.90.
s=SIP Media Capabilities.
c=IN IP4 68.80.201.101.
t=0 0.
m=audio 35230 RTP/AVP 18 101.
a=rtpmap:18 G729/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sendrecv.
a=nortpproxy:yes.
###
U 2004/11/17 14:59:59.215974 216.229.127.60:5060 -> 68.80.201.101:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 68.80.201.101;branch=z9hG4bK6628.1a8491a7.0.
Via: SIP/2.0/UDP
192.168.0.83;rport=1063;received=68.80.200.100;branch=z9hG4bK20ba9258b87f741d.
To: <sip:14075551212@sip.mycompany.com;user=phone>;tag=0bf5212d.
From: "Paul (1002)"
<sip:9990010001@sip.mycompany.com;user=phone>;tag=4edd147cdac0025f.
Call-ID: aa31202374f54793(a)192.168.0.83.
CSeq: 12010 INVITE.
Contact: sip:4075551212@216.229.50.90:4060.
Record-Route: <sip:216.229.127.60:5060;lr>.
Record-Route: <sip:68.80.201.101;ftag=4edd147cdac0025f;lr=on>.
Accept: multipart/mixed, application/sdp, application/isup, application/dtmf,
application/dtmf-relay.
Allow: OPTIONS, INVITE, CANCEL, ACK, BYE, PRACK, INFO.
Supported: timer.
Content-Disposition: session;handling=required.
Content-Type: application/sdp.
Session-Expires: 240;refresher=uas.
Content-Length: 225.
.
v=0.
o=Sonus_UAC 57 25508 IN IP4 216.229.50.90.
s=SIP Media Capabilities.
c=IN IP4 216.229.118.100.
t=0 0.
m=audio 18212 RTP/AVP 18 101.
a=rtpmap:18 G729/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sendrecv.
#
U 2004/11/17 14:59:59.216728 68.80.201.101:5060 -> 68.80.200.100:1063
SIP/2.0 200 OK.
Via: SIP/2.0/UDP
192.168.0.83;rport=1063;received=68.80.200.100;branch=z9hG4bK20ba9258b87f741d.
To: <sip:14075551212@sip.mycompany.com;user=phone>;tag=0bf5212d.
From: "Paul (1002)"
<sip:9990010001@sip.mycompany.com;user=phone>;tag=4edd147cdac0025f.
Call-ID: aa31202374f54793(a)192.168.0.83.
CSeq: 12010 INVITE.
Contact: sip:4075551212@216.229.127.60:5060.
Record-Route: <sip:216.229.127.60:5060;lr>.
Record-Route: <sip:68.80.201.101;ftag=4edd147cdac0025f;lr=on>.
Accept: multipart/mixed, application/sdp, application/isup, application/dtmf,
application/dtmf-relay.
Allow: OPTIONS, INVITE, CANCEL, ACK, BYE, PRACK, INFO.
Supported: timer.
Content-Disposition: session;handling=required.
Content-Type: application/sdp.
Session-Expires: 240;refresher=uas.
Content-Length: 241.
.
v=0.
o=Sonus_UAC 57 25508 IN IP4 216.229.50.90.
s=SIP Media Capabilities.
c=IN IP4 68.80.201.101.
t=0 0.
m=audio 35230 RTP/AVP 18 101.
a=rtpmap:18 G729/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sendrecv.
a=nortpproxy:yes.
#
U 2004/11/17 14:59:59.255342 68.80.200.100:1063 -> 68.80.201.101:5060
ACK sip:4075551212@216.229.127.60:5060 SIP/2.0.
Via: SIP/2.0/UDP 192.168.0.83;branch=z9hG4bKe15e695c98bd9cb0.
Route: <sip:68.80.201.101;ftag=4edd147cdac0025f;lr=on>.
Route: <sip:216.229.127.60:5060;lr>.
From: "Paul (1002)"
<sip:9990010001@sip.mycompany.com;user=phone>;tag=4edd147cdac0025f.
To: <sip:14075551212@sip.mycompany.com;user=phone>;tag=0bf5212d.
Contact: <sip:9990010001@192.168.0.83;user=phone>.
Proxy-Authorization: DIGEST username="9990010001",
realm="sip.mycompany.com", algorithm=MD5,
uri="sip:4075551212@216.229.127.60:5060",
nonce="419baee2139477c22db6d0ec9a47b23003512431",
response="7b83435f78bc60f5ced80304c7d54093".
Call-ID: aa31202374f54793(a)192.168.0.83.
CSeq: 12010 ACK.
User-Agent: Grandstream BT100 1.0.5.11.
Max-Forwards: 70.
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE.
Content-Length: 0.
.
#
U 2004/11/17 14:59:59.255737 68.80.201.101:5060 -> 216.229.127.60:5060
ACK sip:216.229.127.60:5060;lr SIP/2.0.
Record-Route: <sip:68.80.201.101;ftag=4edd147cdac0025f;lr=on>.
Via: SIP/2.0/UDP 68.80.201.101;branch=0.
Via: SIP/2.0/UDP
192.168.0.83;rport=1063;received=68.80.200.100;branch=z9hG4bKe15e695c98bd9cb0.
From: "Paul (1002)"
<sip:9990010001@sip.mycompany.com;user=phone>;tag=4edd147cdac0025f.
To: <sip:14075551212@sip.mycompany.com;user=phone>;tag=0bf5212d.
Contact: <sip:9990010001@68.80.200.100:1063;user=phone>.
Proxy-Authorization: DIGEST username="9990010001",
realm="sip.mycompany.com", algorithm=MD5,
uri="sip:4075551212@216.229.127.60:5060",
nonce="419baee2139477c22db6d0ec9a47b23003512431",
response="7b83435f78bc60f5ced80304c7d54093".
Call-ID: aa31202374f54793(a)192.168.0.83.
CSeq: 12010 ACK.
User-Agent: Grandstream BT100 1.0.5.11.
Max-Forwards: 16.
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE.
Content-Length: 0.
.
#
U 2004/11/17 14:59:59.704902 216.229.127.60:5060 -> 68.80.201.101:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 68.80.201.101;branch=z9hG4bK6628.1a8491a7.0.
Via: SIP/2.0/UDP
192.168.0.83;rport=1063;received=68.80.200.100;branch=z9hG4bK20ba9258b87f741d.
To: <sip:14075551212@sip.mycompany.com;user=phone>;tag=0bf5212d.
From: "Paul (1002)"
<sip:9990010001@216.229.127.60;user=phone>;tag=4edd147cdac0025f.
Call-ID: aa31202374f54793(a)192.168.0.83.
CSeq: 12010 INVITE.
Contact: sip:4075551212@216.229.50.90:4060.
Record-Route: <sip:216.229.127.60:5060;lr>.
Record-Route: <sip:68.80.201.101;ftag=4edd147cdac0025f;lr=on>.
Accept: multipart/mixed, application/sdp, application/isup, application/dtmf,
application/dtmf-relay.
Allow: OPTIONS, INVITE, CANCEL, ACK, BYE, PRACK, INFO.
Supported: timer.
Content-Disposition: session;handling=required.
Content-Type: application/sdp.
Session-Expires: 240;refresher=uas.
Content-Length: 225.
.
v=0.
o=Sonus_UAC 57 25508 IN IP4 216.229.50.90.
s=SIP Media Capabilities.
c=IN IP4 216.229.118.100.
t=0 0.
m=audio 18212 RTP/AVP 18 101.
a=rtpmap:18 G729/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sendrecv.
#
U 2004/11/17 14:59:59.705831 68.80.201.101:5060 -> 68.80.200.100:1063
SIP/2.0 200 OK.
Via: SIP/2.0/UDP
192.168.0.83;rport=1063;received=68.80.200.100;branch=z9hG4bK20ba9258b87f741d.
To: <sip:14075551212@sip.mycompany.com;user=phone>;tag=0bf5212d.
From: "Paul (1002)"
<sip:9990010001@216.229.127.60;user=phone>;tag=4edd147cdac0025f.
Call-ID: aa31202374f54793(a)192.168.0.83.
CSeq: 12010 INVITE.
Contact: sip:4075551212@216.229.127.60:5060.
Record-Route: <sip:216.229.127.60:5060;lr>.
Record-Route: <sip:68.80.201.101;ftag=4edd147cdac0025f;lr=on>.
Accept: multipart/mixed, application/sdp, application/isup, application/dtmf,
application/dtmf-relay.
Allow: OPTIONS, INVITE, CANCEL, ACK, BYE, PRACK, INFO.
Supported: timer.
Content-Disposition: session;handling=required.
Content-Type: application/sdp.
Session-Expires: 240;refresher=uas.
Content-Length: 241.
.
v=0.
o=Sonus_UAC 57 25508 IN IP4 216.229.50.90.
s=SIP Media Capabilities.
c=IN IP4 68.80.201.101.
t=0 0.
m=audio 35230 RTP/AVP 18 101.
a=rtpmap:18 G729/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sendrecv.
a=nortpproxy:yes.
#
U 2004/11/17 14:59:59.715379 68.80.200.100:1063 -> 68.80.201.101:5060
ACK sip:4075551212@216.229.127.60:5060 SIP/2.0.
Via: SIP/2.0/UDP 192.168.0.83;branch=z9hG4bK5d4c53e96e01ccae.
Route: <sip:68.80.201.101;ftag=4edd147cdac0025f;lr=on>.
Route: <sip:216.229.127.60:5060;lr>.
From: "Paul (1002)"
<sip:9990010001@sip.mycompany.com;user=phone>;tag=4edd147cdac0025f.
To: <sip:14075551212@sip.mycompany.com;user=phone>;tag=0bf5212d.
Contact: <sip:9990010001@192.168.0.83;user=phone>.
Proxy-Authorization: DIGEST username="9990010001",
realm="sip.mycompany.com", algorithm=MD5,
uri="sip:4075551212@216.229.127.60:5060",
nonce="419baee2139477c22db6d0ec9a47b23003512431",
response="7b83435f78bc60f5ced80304c7d54093".
Call-ID: aa31202374f54793(a)192.168.0.83.
CSeq: 12010 ACK.
User-Agent: Grandstream BT100 1.0.5.11.
Max-Forwards: 70.
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE.
Content-Length: 0.
.
#
U 2004/11/17 14:59:59.716010 68.80.201.101:5060 -> 216.229.127.60:5060
ACK sip:216.229.127.60:5060;lr SIP/2.0.
Record-Route: <sip:68.80.201.101;ftag=4edd147cdac0025f;lr=on>.
Via: SIP/2.0/UDP 68.80.201.101;branch=0.
Via: SIP/2.0/UDP
192.168.0.83;rport=1063;received=68.80.200.100;branch=z9hG4bK5d4c53e96e01ccae.
From: "Paul (1002)"
<sip:9990010001@sip.mycompany.com;user=phone>;tag=4edd147cdac0025f.
To: <sip:14075551212@sip.mycompany.com;user=phone>;tag=0bf5212d.
Contact: <sip:9990010001@68.80.200.100:1063;user=phone>.
Proxy-Authorization: DIGEST username="9990010001",
realm="sip.mycompany.com", algorithm=MD5,
uri="sip:4075551212@216.229.127.60:5060",
nonce="419baee2139477c22db6d0ec9a47b23003512431",
response="7b83435f78bc60f5ced80304c7d54093".
Call-ID: aa31202374f54793(a)192.168.0.83.
CSeq: 12010 ACK.
User-Agent: Grandstream BT100 1.0.5.11.
Max-Forwards: 16.
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE.
Content-Length: 0.
.
#
U 2004/11/17 15:00:00.705535 216.229.127.60:5060 -> 68.80.201.101:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 68.80.201.101;branch=z9hG4bK6628.1a8491a7.0.
Via: SIP/2.0/UDP
192.168.0.83;rport=1063;received=68.80.200.100;branch=z9hG4bK20ba9258b87f741d.
To: <sip:14075551212@sip.mycompany.com;user=phone>;tag=0bf5212d.
From: "Paul (1002)"
<sip:9990010001@216.229.127.60;user=phone>;tag=4edd147cdac0025f.
Call-ID: aa31202374f54793(a)192.168.0.83.
CSeq: 12010 INVITE.
Contact: sip:4075551212@216.229.50.90:4060.
Record-Route: <sip:216.229.127.60:5060;lr>.
Record-Route: <sip:68.80.201.101;ftag=4edd147cdac0025f;lr=on>.
Accept: multipart/mixed, application/sdp, application/isup, application/dtmf,
application/dtmf-relay.
Allow: OPTIONS, INVITE, CANCEL, ACK, BYE, PRACK, INFO.
Supported: timer.
Content-Disposition: session;handling=required.
Content-Type: application/sdp.
Session-Expires: 240;refresher=uas.
Content-Length: 225.
.
v=0.
o=Sonus_UAC 57 25508 IN IP4 216.229.50.90.
s=SIP Media Capabilities.
c=IN IP4 216.229.118.100.
t=0 0.
m=audio 18212 RTP/AVP 18 101.
a=rtpmap:18 G729/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sendrecv.
#
U 2004/11/17 15:00:00.706510 68.80.201.101:5060 -> 68.80.200.100:1063
SIP/2.0 200 OK.
Via: SIP/2.0/UDP
192.168.0.83;rport=1063;received=68.80.200.100;branch=z9hG4bK20ba9258b87f741d.
To: <sip:14075551212@sip.mycompany.com;user=phone>;tag=0bf5212d.
From: "Paul (1002)"
<sip:9990010001@216.229.127.60;user=phone>;tag=4edd147cdac0025f.
Call-ID: aa31202374f54793(a)192.168.0.83.
CSeq: 12010 INVITE.
Contact: sip:4075551212@216.229.127.60:5060.
Record-Route: <sip:216.229.127.60:5060;lr>.
Record-Route: <sip:68.80.201.101;ftag=4edd147cdac0025f;lr=on>.
Accept: multipart/mixed, application/sdp, application/isup, application/dtmf,
application/dtmf-relay.
Allow: OPTIONS, INVITE, CANCEL, ACK, BYE, PRACK, INFO.
Supported: timer.
Content-Disposition: session;handling=required.
Content-Type: application/sdp.
Session-Expires: 240;refresher=uas.
Content-Length: 241.
.
v=0.
o=Sonus_UAC 57 25508 IN IP4 216.229.50.90.
s=SIP Media Capabilities.
c=IN IP4 68.80.201.101.
t=0 0.
m=audio 35230 RTP/AVP 18 101.
a=rtpmap:18 G729/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sendrecv.
a=nortpproxy:yes.
#
U 2004/11/17 15:00:00.716029 68.80.200.100:1063 -> 68.80.201.101:5060
ACK sip:4075551212@216.229.127.60:5060 SIP/2.0.
Via: SIP/2.0/UDP 192.168.0.83;branch=z9hG4bK71e568739c660904.
Route: <sip:68.80.201.101;ftag=4edd147cdac0025f;lr=on>.
Route: <sip:216.229.127.60:5060;lr>.
From: "Paul (1002)"
<sip:9990010001@sip.mycompany.com;user=phone>;tag=4edd147cdac0025f.
To: <sip:14075551212@sip.mycompany.com;user=phone>;tag=0bf5212d.
Contact: <sip:9990010001@192.168.0.83;user=phone>.
Proxy-Authorization: DIGEST username="9990010001",
realm="sip.mycompany.com", algorithm=MD5,
uri="sip:4075551212@216.229.127.60:5060",
nonce="419baee2139477c22db6d0ec9a47b23003512431",
response="7b83435f78bc60f5ced80304c7d54093".
Call-ID: aa31202374f54793(a)192.168.0.83.
CSeq: 12010 ACK.
User-Agent: Grandstream BT100 1.0.5.11.
Max-Forwards: 70.
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE.
Content-Length: 0.
.
#
U 2004/11/17 15:00:00.716669 68.80.201.101:5060 -> 216.229.127.60:5060
ACK sip:216.229.127.60:5060;lr SIP/2.0.
Record-Route: <sip:68.80.201.101;ftag=4edd147cdac0025f;lr=on>.
Via: SIP/2.0/UDP 68.80.201.101;branch=0.
Via: SIP/2.0/UDP
192.168.0.83;rport=1063;received=68.80.200.100;branch=z9hG4bK71e568739c660904.
From: "Paul (1002)"
<sip:9990010001@sip.mycompany.com;user=phone>;tag=4edd147cdac0025f.
To: <sip:14075551212@sip.mycompany.com;user=phone>;tag=0bf5212d.
Contact: <sip:9990010001@68.80.200.100:1063;user=phone>.
Proxy-Authorization: DIGEST username="9990010001",
realm="sip.mycompany.com", algorithm=MD5,
uri="sip:4075551212@216.229.127.60:5060",
nonce="419baee2139477c22db6d0ec9a47b23003512431",
response="7b83435f78bc60f5ced80304c7d54093".
Call-ID: aa31202374f54793(a)192.168.0.83.
CSeq: 12010 ACK.
User-Agent: Grandstream BT100 1.0.5.11.
Max-Forwards: 16.
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE.
Content-Length: 0.
.
#
U 2004/11/17 15:00:02.704680 216.229.127.60:5060 -> 68.80.201.101:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 68.80.201.101;branch=z9hG4bK6628.1a8491a7.0.
Via: SIP/2.0/UDP
192.168.0.83;rport=1063;received=68.80.200.100;branch=z9hG4bK20ba9258b87f741d.
To: <sip:14075551212@sip.mycompany.com;user=phone>;tag=0bf5212d.
From: "Paul (1002)"
<sip:9990010001@216.229.127.60;user=phone>;tag=4edd147cdac0025f.
Call-ID: aa31202374f54793(a)192.168.0.83.
CSeq: 12010 INVITE.
Contact: sip:4075551212@216.229.50.90:4060.
Record-Route: <sip:216.229.127.60:5060;lr>.
Record-Route: <sip:68.80.201.101;ftag=4edd147cdac0025f;lr=on>.
Accept: multipart/mixed, application/sdp, application/isup, application/dtmf,
application/dtmf-relay.
Allow: OPTIONS, INVITE, CANCEL, ACK, BYE, PRACK, INFO.
Supported: timer.
Content-Disposition: session;handling=required.
Content-Type: application/sdp.
Session-Expires: 240;refresher=uas.
Content-Length: 225.
.
v=0.
o=Sonus_UAC 57 25508 IN IP4 216.229.50.90.
s=SIP Media Capabilities.
c=IN IP4 216.229.118.100.
t=0 0.
m=audio 18212 RTP/AVP 18 101.
a=rtpmap:18 G729/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sendrecv.
#
U 2004/11/17 15:00:02.705726 68.80.201.101:5060 -> 68.80.200.100:1063
SIP/2.0 200 OK.
Via: SIP/2.0/UDP
192.168.0.83;rport=1063;received=68.80.200.100;branch=z9hG4bK20ba9258b87f741d.
To: <sip:14075551212@sip.mycompany.com;user=phone>;tag=0bf5212d.
From: "Paul (1002)"
<sip:9990010001@216.229.127.60;user=phone>;tag=4edd147cdac0025f.
Call-ID: aa31202374f54793(a)192.168.0.83.
CSeq: 12010 INVITE.
Contact: sip:4075551212@216.229.127.60:5060.
Record-Route: <sip:216.229.127.60:5060;lr>.
Record-Route: <sip:68.80.201.101;ftag=4edd147cdac0025f;lr=on>.
Accept: multipart/mixed, application/sdp, application/isup, application/dtmf,
application/dtmf-relay.
Allow: OPTIONS, INVITE, CANCEL, ACK, BYE, PRACK, INFO.
Supported: timer.
Content-Disposition: session;handling=required.
Content-Type: application/sdp.
Session-Expires: 240;refresher=uas.
Content-Length: 241.
.
v=0.
o=Sonus_UAC 57 25508 IN IP4 216.229.50.90.
s=SIP Media Capabilities.
c=IN IP4 68.80.201.101.
t=0 0.
m=audio 35230 RTP/AVP 18 101.
a=rtpmap:18 G729/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sendrecv.
a=nortpproxy:yes.
#
U 2004/11/17 15:00:02.715249 68.80.200.100:1063 -> 68.80.201.101:5060
ACK sip:4075551212@216.229.127.60:5060 SIP/2.0.
Via: SIP/2.0/UDP 192.168.0.83;branch=z9hG4bK224b67d8d70043ad.
Route: <sip:68.80.201.101;ftag=4edd147cdac0025f;lr=on>.
Route: <sip:216.229.127.60:5060;lr>.
From: "Paul (1002)"
<sip:9990010001@sip.mycompany.com;user=phone>;tag=4edd147cdac0025f.
To: <sip:14075551212@sip.mycompany.com;user=phone>;tag=0bf5212d.
Contact: <sip:9990010001@192.168.0.83;user=phone>.
Proxy-Authorization: DIGEST username="9990010001",
realm="sip.mycompany.com", algorithm=MD5,
uri="sip:4075551212@216.229.127.60:5060",
nonce="419baee2139477c22db6d0ec9a47b23003512431",
response="7b83435f78bc60f5ced80304c7d54093".
Call-ID: aa31202374f54793(a)192.168.0.83.
CSeq: 12010 ACK.
User-Agent: Grandstream BT100 1.0.5.11.
Max-Forwards: 70.
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE.
Content-Length: 0.
.
#
U 2004/11/17 15:00:02.715905 68.80.201.101:5060 -> 216.229.127.60:5060
ACK sip:216.229.127.60:5060;lr SIP/2.0.
Record-Route: <sip:68.80.201.101;ftag=4edd147cdac0025f;lr=on>.
Via: SIP/2.0/UDP 68.80.201.101;branch=0.
Via: SIP/2.0/UDP
192.168.0.83;rport=1063;received=68.80.200.100;branch=z9hG4bK224b67d8d70043ad.
From: "Paul (1002)"
<sip:9990010001@sip.mycompany.com;user=phone>;tag=4edd147cdac0025f.
To: <sip:14075551212@sip.mycompany.com;user=phone>;tag=0bf5212d.
Contact: <sip:9990010001@68.80.200.100:1063;user=phone>.
Proxy-Authorization: DIGEST username="9990010001",
realm="sip.mycompany.com", algorithm=MD5,
uri="sip:4075551212@216.229.127.60:5060",
nonce="419baee2139477c22db6d0ec9a47b23003512431",
response="7b83435f78bc60f5ced80304c7d54093".
Call-ID: aa31202374f54793(a)192.168.0.83.
CSeq: 12010 ACK.
User-Agent: Grandstream BT100 1.0.5.11.
Max-Forwards: 16.
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE.
Content-Length: 0.
.
#
U 2004/11/17 15:00:06.705724 216.229.127.60:5060 -> 68.80.201.101:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 68.80.201.101;branch=z9hG4bK6628.1a8491a7.0.
Via: SIP/2.0/UDP
192.168.0.83;rport=1063;received=68.80.200.100;branch=z9hG4bK20ba9258b87f741d.
To: <sip:14075551212@sip.mycompany.com;user=phone>;tag=0bf5212d.
From: "Paul (1002)"
<sip:9990010001@216.229.127.60;user=phone>;tag=4edd147cdac0025f.
Call-ID: aa31202374f54793(a)192.168.0.83.
CSeq: 12010 INVITE.
Contact: sip:4075551212@216.229.50.90:4060.
Record-Route: <sip:216.229.127.60:5060;lr>.
Record-Route: <sip:68.80.201.101;ftag=4edd147cdac0025f;lr=on>.
Accept: multipart/mixed, application/sdp, application/isup, application/dtmf,
application/dtmf-relay.
Allow: OPTIONS, INVITE, CANCEL, ACK, BYE, PRACK, INFO.
Supported: timer.
Content-Disposition: session;handling=required.
Content-Type: application/sdp.
Session-Expires: 240;refresher=uas.
Content-Length: 225.
.
v=0.
o=Sonus_UAC 57 25508 IN IP4 216.229.50.90.
s=SIP Media Capabilities.
c=IN IP4 216.229.118.100.
t=0 0.
m=audio 18212 RTP/AVP 18 101.
a=rtpmap:18 G729/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sendrecv.
#
U 2004/11/17 15:00:06.705948 68.80.201.101:5060 -> 68.80.200.100:1063
SIP/2.0 200 OK.
Via: SIP/2.0/UDP
192.168.0.83;rport=1063;received=68.80.200.100;branch=z9hG4bK20ba9258b87f741d.
To: <sip:14075551212@sip.mycompany.com;user=phone>;tag=0bf5212d.
From: "Paul (1002)"
<sip:9990010001@216.229.127.60;user=phone>;tag=4edd147cdac0025f.
Call-ID: aa31202374f54793(a)192.168.0.83.
CSeq: 12010 INVITE.
Contact: sip:4075551212@216.229.50.90:4060.
Record-Route: <sip:216.229.127.60:5060;lr>.
Record-Route: <sip:68.80.201.101;ftag=4edd147cdac0025f;lr=on>.
Accept: multipart/mixed, application/sdp, application/isup, application/dtmf,
application/dtmf-relay.
Allow: OPTIONS, INVITE, CANCEL, ACK, BYE, PRACK, INFO.
Supported: timer.
Content-Disposition: session;handling=required.
Content-Type: application/sdp.
Session-Expires: 240;refresher=uas.
Content-Length: 225.
.
v=0.
o=Sonus_UAC 57 25508 IN IP4 216.229.50.90.
s=SIP Media Capabilities.
c=IN IP4 216.229.118.100.
t=0 0.
m=audio 18212 RTP/AVP 18 101.
a=rtpmap:18 G729/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sendrecv.
#
#
#########
#
U 2004/11/17 15:00:10.704340 216.229.127.60:5060 -> 68.80.201.101:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 68.80.201.101;branch=z9hG4bK6628.1a8491a7.0.
Via: SIP/2.0/UDP
192.168.0.83;rport=1063;received=68.80.200.100;branch=z9hG4bK20ba9258b87f741d.
To: <sip:14075551212@sip.mycompany.com;user=phone>;tag=0bf5212d.
From: "Paul (1002)"
<sip:9990010001@216.229.127.60;user=phone>;tag=4edd147cdac0025f.
Call-ID: aa31202374f54793(a)192.168.0.83.
CSeq: 12010 INVITE.
Contact: sip:4075551212@216.229.50.90:4060.
Record-Route: <sip:216.229.127.60:5060;lr>.
Record-Route: <sip:68.80.201.101;ftag=4edd147cdac0025f;lr=on>.
Accept: multipart/mixed, application/sdp, application/isup, application/dtmf,
application/dtmf-relay.
Allow: OPTIONS, INVITE, CANCEL, ACK, BYE, PRACK, INFO.
Supported: timer.
Content-Disposition: session;handling=required.
Content-Type: application/sdp.
Session-Expires: 240;refresher=uas.
Content-Length: 225.
.
v=0.
o=Sonus_UAC 57 25508 IN IP4 216.229.50.90.
s=SIP Media Capabilities.
c=IN IP4 216.229.118.100.
t=0 0.
m=audio 18212 RTP/AVP 18 101.
a=rtpmap:18 G729/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sendrecv.
#
U 2004/11/17 15:00:10.704547 68.80.201.101:5060 -> 68.80.200.100:1063
SIP/2.0 200 OK.
Via: SIP/2.0/UDP
192.168.0.83;rport=1063;received=68.80.200.100;branch=z9hG4bK20ba9258b87f741d.
To: <sip:14075551212@sip.mycompany.com;user=phone>;tag=0bf5212d.
From: "Paul (1002)"
<sip:9990010001@216.229.127.60;user=phone>;tag=4edd147cdac0025f.
Call-ID: aa31202374f54793(a)192.168.0.83.
CSeq: 12010 INVITE.
Contact: sip:4075551212@216.229.50.90:4060.
Record-Route: <sip:216.229.127.60:5060;lr>.
Record-Route: <sip:68.80.201.101;ftag=4edd147cdac0025f;lr=on>.
Accept: multipart/mixed, application/sdp, application/isup, application/dtmf,
application/dtmf-relay.
Allow: OPTIONS, INVITE, CANCEL, ACK, BYE, PRACK, INFO.
Supported: timer.
Content-Disposition: session;handling=required.
Content-Type: application/sdp.
Session-Expires: 240;refresher=uas.
Content-Length: 225.
.
v=0.
o=Sonus_UAC 57 25508 IN IP4 216.229.50.90.
s=SIP Media Capabilities.
c=IN IP4 216.229.118.100.
t=0 0.
m=audio 18212 RTP/AVP 18 101.
a=rtpmap:18 G729/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sendrecv.
18 Nov
18 Nov
8:52 a.m.
U 2004/11/17 14:59:59.255342 68.80.200.100:1063 ->
68.80.201.101:5060
ACK sip:4075551212@216.229.127.60:5060 SIP/2.0.
Via: SIP/2.0/UDP 192.168.0.83;branch=z9hG4bKe15e695c98bd9cb0.
Route: <sip:68.80.201.101;ftag=4edd147cdac0025f;lr=on>.
Route: <sip:216.229.127.60:5060;lr>.
From: "Paul (1002)"
<sip:9990010001@sip.mycompany.com;user=phone>;tag=4edd147cdac0025f.
To: <sip:14075551212@sip.mycompany.com;user=phone>;tag=0bf5212d.
Contact: <sip:9990010001@192.168.0.83;user=phone>.
Proxy-Authorization: DIGEST username="9990010001",
realm="sip.mycompany.com", algorithm=MD5,
uri="sip:4075551212@216.229.127.60:5060",
nonce="419baee2139477c22db6d0ec9a47b23003512431",
response="7b83435f78bc60f5ced80304c7d54093".
Call-ID: aa31202374f54793(a)192.168.0.83.
CSeq: 12010 ACK.
User-Agent: Grandstream BT100 1.0.5.11.
Max-Forwards: 70.
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE.
Content-Length: 0.
.
#
U 2004/11/17 14:59:59.255737 68.80.201.101:5060 -> 216.229.127.60:5060
ACK sip:216.229.127.60:5060;lr SIP/2.0.
Record-Route: <sip:68.80.201.101;ftag=4edd147cdac0025f;lr=on>.
Via: SIP/2.0/UDP 68.80.201.101;branch=0.
Via: SIP/2.0/UDP
192.168.0.83;rport=1063;received=68.80.200.100;branch=z9hG4bKe15e695c98bd9cb0.
From: "Paul (1002)"
<sip:9990010001@sip.mycompany.com;user=phone>;tag=4edd147cdac0025f.
To: <sip:14075551212@sip.mycompany.com;user=phone>;tag=0bf5212d.
Contact: <sip:9990010001@68.80.200.100:1063;user=phone>.
Proxy-Authorization: DIGEST username="9990010001",
realm="sip.mycompany.com", algorithm=MD5,
uri="sip:4075551212@216.229.127.60:5060",
nonce="419baee2139477c22db6d0ec9a47b23003512431",
response="7b83435f78bc60f5ced80304c7d54093".
Call-ID: aa31202374f54793(a)192.168.0.83.
CSeq: 12010 ACK.
User-Agent: Grandstream BT100 1.0.5.11.
Max-Forwards: 16.
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE.
Content-Length: 0.
.
#
I guess, you are doing record-route for ACK also. IMO, you should not
add record route headers for ACK.
I am not sure why Route header is completely removed from the request.
Proxy is supposed to delete only the top route value.
-Ranga
9:13 a.m.
Ranga,
Yes, in my ser.cfg script I have
if (!method=="REGISTER") record-route();
I think there is plenty of confusion in the serusers list about wheather or not this is
correct.
Most of the example ser.cfg files I've seen, including those that sip with ser, show
this rather
than
if (!method=="INVITE") record-route();
Can anyone say which method is more RFC3261 compliant?
Also, I can't say wheather or not changing my ser.cfg to the latter will fix the
problems that our
PSTN provider has with our SIP dialogs. I'll find that out in the morning.
Regards,
Paul
--- Ranga <rangarao.v(a)gmail.com> wrote:
__________________________________
Do you Yahoo!?
The all-new My Yahoo! - Get yours free!
http://my.yahoo.com
U 2004/11/17
14:59:59.255342 68.80.200.100:1063 -> 68.80.201.101:5060
ACK sip:4075551212@216.229.127.60:5060 SIP/2.0.
Via: SIP/2.0/UDP 192.168.0.83;branch=z9hG4bKe15e695c98bd9cb0.
Route: <sip:68.80.201.101;ftag=4edd147cdac0025f;lr=on>.
Route: <sip:216.229.127.60:5060;lr>.
From: "Paul (1002)"
<sip:9990010001@sip.mycompany.com;user=phone>;tag=4edd147cdac0025f.
To: <sip:14075551212@sip.mycompany.com;user=phone>;tag=0bf5212d.
Contact: <sip:9990010001@192.168.0.83;user=phone>.
Proxy-Authorization: DIGEST username="9990010001",
realm="sip.mycompany.com", algorithm=MD5,
uri="sip:4075551212@216.229.127.60:5060",
nonce="419baee2139477c22db6d0ec9a47b23003512431",
response="7b83435f78bc60f5ced80304c7d54093".
Call-ID: aa31202374f54793(a)192.168.0.83.
CSeq: 12010 ACK.
User-Agent: Grandstream BT100 1.0.5.11.
Max-Forwards: 70.
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE.
Content-Length: 0.
.
#
U 2004/11/17 14:59:59.255737 68.80.201.101:5060 -> 216.229.127.60:5060
ACK sip:216.229.127.60:5060;lr SIP/2.0.
Record-Route: <sip:68.80.201.101;ftag=4edd147cdac0025f;lr=on>.
Via: SIP/2.0/UDP 68.80.201.101;branch=0.
Via: SIP/2.0/UDP
192.168.0.83;rport=1063;received=68.80.200.100;branch=z9hG4bKe15e695c98bd9cb0.
From: "Paul (1002)"
<sip:9990010001@sip.mycompany.com;user=phone>;tag=4edd147cdac0025f.
To: <sip:14075551212@sip.mycompany.com;user=phone>;tag=0bf5212d.
Contact: <sip:9990010001@68.80.200.100:1063;user=phone>.
Proxy-Authorization: DIGEST username="9990010001",
realm="sip.mycompany.com", algorithm=MD5,
uri="sip:4075551212@216.229.127.60:5060",
nonce="419baee2139477c22db6d0ec9a47b23003512431",
response="7b83435f78bc60f5ced80304c7d54093".
Call-ID: aa31202374f54793(a)192.168.0.83.
CSeq: 12010 ACK.
User-Agent: Grandstream BT100 1.0.5.11.
Max-Forwards: 16.
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE.
Content-Length: 0.
.
#
I guess, you are doing record-route for ACK also. IMO, you should not
add record route headers for ACK.
I am not sure why Route header is completely removed from the request.
Proxy is supposed to delete only the top route value.
-Ranga
9:43 a.m.
I may have stumbled upon the root of the problem and a solution.
In my ser.cfg I have
if (!method=="INVITE") record-route();
And my phone is a Grandstream BT100. I did not have anything in my "Outbound
Proxy" field because
I assumed that omitting this field would default the phone to use the ser proxy as the
outbound
proxy.
I now see that this is not the case. When my outbound proxy field was left blank my ACK
looked
like this:
NOTE: These ACKs are originating at my ser proxy and being sent to the Sonus equipment.
U 2004/11/18 02:19:05.789518 66.40.100.99:5060 -> 216.12.18.98:5060
ACK sip:216.12.18.98:5060;lr SIP/2.0.
Via: SIP/2.0/UDP 66.40.100.99;branch=0.
Via: SIP/2.0/UDP
172.16.1.34;rport=5060;received=66.90.50.230;branch=z9hG4bK0cf6f59ad9cff263.
From: "Paul Hazlett"
<sip:9990010005@sip.mycompany.com;user=phone>;tag=153be9cfe0227ba6.
To: <sip:14075551212@sip.mycompany.com;user=phone>;tag=0bc2b91b.
Contact: <sip:9990010005@66.90.50.230:5060;user=phone>.
Proxy-Authorization: DIGEST username="9990010005",
realm="sip.mycompany.com", algorithm=MD5,
uri="sip:4075551212@216.12.18.98:5060",
nonce="419c4e0f62b492f993917bb283d89431c978b522",
response="4dd7cb5d0e06a50c24d3a4848f57f1ad".
Call-ID: 2fe29b9654abc94f(a)172.16.1.34.
CSeq: 16138 ACK.
User-Agent: Grandstream BT100 1.0.5.11.
Max-Forwards: 16.
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE.
Content-Length: 0.
Now with my outbound proxy field set to the same value as my sip proxy I get this ACK --
notice
the
Route: header??? The Sonus equipment now seems to be happy.
U 2004/11/18 02:29:47.505321 66.40.100.99:5060 -> 216.12.18.98:5060
ACK sip:4075551212@216.229.118.76:4060 SIP/2.0.
Via: SIP/2.0/UDP 66.40.100.99;branch=0.
Via: SIP/2.0/UDP 66.90.50.230;branch=z9hG4bKd14abbdd801339a6.
Route: <sip:216.12.18.98:5060;lr>.
From: "Paul Hazlett"
<sip:9990010005@sip.mycompany.com;user=phone>;tag=343704cb47bd81b2.
To: <sip:14075551212@sip.mycompany.com;user=phone>;tag=06c57864.
Contact: <sip:9990010005@66.90.50.230;user=phone>.
Proxy-Authorization: DIGEST username="9990010005",
realm="sip.mycompany.com", algorithm=MD5,
uri="sip:4075551212@216.229.118.76:4060",
nonce="419c50915314f17ec7beef27579c4fc2ccfba7ed",
response="6cde7461dfde9e0a7327fa1a20f47fb0".
Call-ID: f211021ad0a34a37(a)172.16.1.34.
CSeq: 3685 ACK.
User-Agent: Grandstream BT100 1.0.5.11.
Max-Forwards: 16.
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE.
Content-Length: 0.
--- Java Rockx <javarockx(a)yahoo.com> wrote:
Ranga,
Yes, in my ser.cfg script I have
if (!method=="REGISTER") record-route();
I think there is plenty of confusion in the serusers list about wheather or not this is
correct.
Most of the example ser.cfg files I've seen, including those that sip with ser, show
this rather
than
if (!method=="INVITE") record-route();
Can anyone say which method is more RFC3261 compliant?
Also, I can't say wheather or not changing my ser.cfg to the latter will fix the
problems that
our
PSTN provider has with our SIP dialogs. I'll find that out in the morning.
Regards,
Paul
--- Ranga <rangarao.v(a)gmail.com> wrote:
__________________________________
Do you Yahoo!?
The all-new My Yahoo! - Get yours free!
http://my.yahoo.com
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
__________________________________
Do you Yahoo!?
Meet the all-new My Yahoo! - Try it today!
http://my.yahoo.com
U
2004/11/17 14:59:59.255342 68.80.200.100:1063 -> 68.80.201.101:5060
ACK sip:4075551212@216.229.127.60:5060 SIP/2.0.
Via: SIP/2.0/UDP 192.168.0.83;branch=z9hG4bKe15e695c98bd9cb0.
Route: <sip:68.80.201.101;ftag=4edd147cdac0025f;lr=on>.
Route: <sip:216.229.127.60:5060;lr>.
From: "Paul (1002)"
<sip:9990010001@sip.mycompany.com;user=phone>;tag=4edd147cdac0025f.
To: <sip:14075551212@sip.mycompany.com;user=phone>;tag=0bf5212d.
Contact: <sip:9990010001@192.168.0.83;user=phone>.
Proxy-Authorization: DIGEST username="9990010001",
realm="sip.mycompany.com", algorithm=MD5,
uri="sip:4075551212@216.229.127.60:5060",
nonce="419baee2139477c22db6d0ec9a47b23003512431",
response="7b83435f78bc60f5ced80304c7d54093".
Call-ID: aa31202374f54793(a)192.168.0.83.
CSeq: 12010 ACK.
User-Agent: Grandstream BT100 1.0.5.11.
Max-Forwards: 70.
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE.
Content-Length: 0.
.
#
U 2004/11/17 14:59:59.255737 68.80.201.101:5060 -> 216.229.127.60:5060
ACK sip:216.229.127.60:5060;lr SIP/2.0.
Record-Route: <sip:68.80.201.101;ftag=4edd147cdac0025f;lr=on>.
Via: SIP/2.0/UDP 68.80.201.101;branch=0.
Via: SIP/2.0/UDP
192.168.0.83;rport=1063;received=68.80.200.100;branch=z9hG4bKe15e695c98bd9cb0.
From: "Paul (1002)"
<sip:9990010001@sip.mycompany.com;user=phone>;tag=4edd147cdac0025f.
To: <sip:14075551212@sip.mycompany.com;user=phone>;tag=0bf5212d.
Contact: <sip:9990010001@68.80.200.100:1063;user=phone>.
Proxy-Authorization: DIGEST username="9990010001",
realm="sip.mycompany.com", algorithm=MD5,
uri="sip:4075551212@216.229.127.60:5060",
nonce="419baee2139477c22db6d0ec9a47b23003512431",
response="7b83435f78bc60f5ced80304c7d54093".
Call-ID: aa31202374f54793(a)192.168.0.83.
CSeq: 12010 ACK.
User-Agent: Grandstream BT100 1.0.5.11.
Max-Forwards: 16.
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE.
Content-Length: 0.
.
#
I guess, you are doing record-route for ACK also. IMO, you should not
add record route headers for ACK.
I am not sure why Route header is completely removed from the request.
Proxy is supposed to delete only the top route value.
-Ranga
9:56 a.m.
Hi All.
One last thing, is I had a STUN server set whereas on earlier attempts I did not have
stun. I'm
using nathelper and rtpproxy so I'm trying to avoid STUN, but it appears for some
reason SER
correctly adds the "Route:" header only when using a STUN server with my
Grandstream phone.
Can anyone shed some light on why this might be?
Regards,
Paul
--- Java Rockx <javarockx(a)yahoo.com> wrote:
I may have stumbled upon the root of the problem and a
solution.
In my ser.cfg I have
if (!method=="INVITE") record-route();
And my phone is a Grandstream BT100. I did not have anything in my "Outbound
Proxy" field
because
I assumed that omitting this field would default the phone to use the ser proxy as the
outbound
proxy.
I now see that this is not the case. When my outbound proxy field was left blank my ACK
looked
like this:
NOTE: These ACKs are originating at my ser proxy and being sent to the Sonus equipment.
U 2004/11/18 02:19:05.789518 66.40.100.99:5060 -> 216.12.18.98:5060
ACK sip:216.12.18.98:5060;lr SIP/2.0.
Via: SIP/2.0/UDP 66.40.100.99;branch=0.
Via: SIP/2.0/UDP
172.16.1.34;rport=5060;received=66.90.50.230;branch=z9hG4bK0cf6f59ad9cff263.
From: "Paul Hazlett"
<sip:9990010005@sip.mycompany.com;user=phone>;tag=153be9cfe0227ba6.
To: <sip:14075551212@sip.mycompany.com;user=phone>;tag=0bc2b91b.
Contact: <sip:9990010005@66.90.50.230:5060;user=phone>.
Proxy-Authorization: DIGEST username="9990010005",
realm="sip.mycompany.com", algorithm=MD5,
uri="sip:4075551212@216.12.18.98:5060",
nonce="419c4e0f62b492f993917bb283d89431c978b522",
response="4dd7cb5d0e06a50c24d3a4848f57f1ad".
Call-ID: 2fe29b9654abc94f(a)172.16.1.34.
CSeq: 16138 ACK.
User-Agent: Grandstream BT100 1.0.5.11.
Max-Forwards: 16.
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE.
Content-Length: 0.
Now with my outbound proxy field set to the same value as my sip proxy I get this ACK --
notice
the
Route: header??? The Sonus equipment now seems to be happy.
U 2004/11/18 02:29:47.505321 66.40.100.99:5060 -> 216.12.18.98:5060
ACK sip:4075551212@216.229.118.76:4060 SIP/2.0.
Via: SIP/2.0/UDP 66.40.100.99;branch=0.
Via: SIP/2.0/UDP 66.90.50.230;branch=z9hG4bKd14abbdd801339a6.
Route: <sip:216.12.18.98:5060;lr>.
From: "Paul Hazlett"
<sip:9990010005@sip.mycompany.com;user=phone>;tag=343704cb47bd81b2.
To: <sip:14075551212@sip.mycompany.com;user=phone>;tag=06c57864.
Contact: <sip:9990010005@66.90.50.230;user=phone>.
Proxy-Authorization: DIGEST username="9990010005",
realm="sip.mycompany.com", algorithm=MD5,
uri="sip:4075551212@216.229.118.76:4060",
nonce="419c50915314f17ec7beef27579c4fc2ccfba7ed",
response="6cde7461dfde9e0a7327fa1a20f47fb0".
Call-ID: f211021ad0a34a37(a)172.16.1.34.
CSeq: 3685 ACK.
User-Agent: Grandstream BT100 1.0.5.11.
Max-Forwards: 16.
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE.
Content-Length: 0.
--- Java Rockx <javarockx(a)yahoo.com> wrote:
__________________________________
Do you Yahoo!?
Meet the all-new My Yahoo! - Try it today!
http://my.yahoo.com
__________________________________
Do you Yahoo!?
The all-new My Yahoo! - Get yours free!
http://my.yahoo.com
Ranga,
Yes, in my ser.cfg script I have
if (!method=="REGISTER") record-route();
I think there is plenty of confusion in the serusers list about wheather or not this is
correct.
Most of the example ser.cfg files I've seen,
including those that sip with ser, show this
rather
than
if (!method=="INVITE") record-route();
Can anyone say which method is more RFC3261 compliant?
Also, I can't say wheather or not changing my ser.cfg to the latter will fix the
problems that
our
PSTN provider has with our SIP dialogs. I'll find that out in the morning.
Regards,
Paul
--- Ranga <rangarao.v(a)gmail.com> wrote:
> > U 2004/11/17 14:59:59.255342 68.80.200.100:1063 -> 68.80.201.101:5060
> > ACK sip:4075551212@216.229.127.60:5060 SIP/2.0.
> > Via: SIP/2.0/UDP 192.168.0.83;branch=z9hG4bKe15e695c98bd9cb0.
> > Route: <sip:68.80.201.101;ftag=4edd147cdac0025f;lr=on>.
> > Route: <sip:216.229.127.60:5060;lr>.
> > From: "Paul (1002)"
<sip:9990010001@sip.mycompany.com;user=phone>;tag=4edd147cdac0025f.
> > To: <sip:14075551212@sip.mycompany.com;user=phone>;tag=0bf5212d.
> > Contact: <sip:9990010001@192.168.0.83;user=phone>.
> > Proxy-Authorization: DIGEST username="9990010001",
realm="sip.mycompany.com",
algorithm=MD5,
>
uri="sip:4075551212@216.229.127.60:5060",
nonce="419baee2139477c22db6d0ec9a47b23003512431",
> response="7b83435f78bc60f5ced80304c7d54093".
> > Call-ID: aa31202374f54793(a)192.168.0.83.
> > CSeq: 12010 ACK.
> > User-Agent: Grandstream BT100 1.0.5.11.
> > Max-Forwards: 70.
> > Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE.
> > Content-Length: 0.
> > .
> > #
> > U 2004/11/17 14:59:59.255737 68.80.201.101:5060 -> 216.229.127.60:5060
> > ACK sip:216.229.127.60:5060;lr SIP/2.0.
> > Record-Route: <sip:68.80.201.101;ftag=4edd147cdac0025f;lr=on>.
> > Via: SIP/2.0/UDP 68.80.201.101;branch=0.
> > Via: SIP/2.0/UDP
> 192.168.0.83;rport=1063;received=68.80.200.100;branch=z9hG4bKe15e695c98bd9cb0.
> > From: "Paul (1002)"
<sip:9990010001@sip.mycompany.com;user=phone>;tag=4edd147cdac0025f.
> > To: <sip:14075551212@sip.mycompany.com;user=phone>;tag=0bf5212d.
> > Contact: <sip:9990010001@68.80.200.100:1063;user=phone>.
> > Proxy-Authorization: DIGEST username="9990010001",
realm="sip.mycompany.com",
algorithm=MD5,
uri="sip:4075551212@216.229.127.60:5060",
nonce="419baee2139477c22db6d0ec9a47b23003512431",
response="7b83435f78bc60f5ced80304c7d54093".
__________________________________
Do you Yahoo!?
The all-new My Yahoo! - Get yours free!
http://my.yahoo.com
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
Call-ID: aa31202374f54793(a)192.168.0.83.
CSeq: 12010 ACK.
User-Agent: Grandstream BT100 1.0.5.11.
Max-Forwards: 16.
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE.
Content-Length: 0.
.
#
I guess, you are doing record-route for ACK also. IMO, you should not
add record route headers for ACK.
I am not sure why Route header is completely removed from the request.
Proxy is supposed to delete only the top route value.
-Ranga
11:48 a.m.
Hi Paul,
If I understand correctly what you are trying to do, I would say the
condition should look like:
if (method=="INVITE") record-route();
^^^^^^^^^^^^^^^^
As INVITE is first request from a dialog, IMHO it must be record-routed
in order to set the route set for all following requests within the dialog.
Best regards,
Marian
Java Rockx wrote:
Ranga,
Yes, in my ser.cfg script I have
if (!method=="REGISTER") record-route();
I think there is plenty of confusion in the serusers list about wheather or not this is
correct.
Most of the example ser.cfg files I've seen, including those that sip with ser, show
this rather
than
if (!method=="INVITE") record-route();
Can anyone say which method is more RFC3261 compliant?
Also, I can't say wheather or not changing my ser.cfg to the latter will fix the
problems that our
PSTN provider has with our SIP dialogs. I'll find that out in the morning.
Regards,
Paul
--- Ranga <rangarao.v(a)gmail.com> wrote:
__________________________________
Do you Yahoo!?
The all-new My Yahoo! - Get yours free!
http://my.yahoo.com
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
--
Voice Sistem
http://www.voice-sistem.ro
U
2004/11/17 14:59:59.255342 68.80.200.100:1063 -> 68.80.201.101:5060
ACK sip:4075551212@216.229.127.60:5060 SIP/2.0.
Via: SIP/2.0/UDP 192.168.0.83;branch=z9hG4bKe15e695c98bd9cb0.
Route: <sip:68.80.201.101;ftag=4edd147cdac0025f;lr=on>.
Route: <sip:216.229.127.60:5060;lr>.
From: "Paul (1002)"
<sip:9990010001@sip.mycompany.com;user=phone>;tag=4edd147cdac0025f.
To: <sip:14075551212@sip.mycompany.com;user=phone>;tag=0bf5212d.
Contact: <sip:9990010001@192.168.0.83;user=phone>.
Proxy-Authorization: DIGEST username="9990010001",
realm="sip.mycompany.com", algorithm=MD5,
uri="sip:4075551212@216.229.127.60:5060",
nonce="419baee2139477c22db6d0ec9a47b23003512431",
response="7b83435f78bc60f5ced80304c7d54093".
Call-ID: aa31202374f54793(a)192.168.0.83.
CSeq: 12010 ACK.
User-Agent: Grandstream BT100 1.0.5.11.
Max-Forwards: 70.
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE.
Content-Length: 0.
.
#
U 2004/11/17 14:59:59.255737 68.80.201.101:5060 -> 216.229.127.60:5060
ACK sip:216.229.127.60:5060;lr SIP/2.0.
Record-Route: <sip:68.80.201.101;ftag=4edd147cdac0025f;lr=on>.
Via: SIP/2.0/UDP 68.80.201.101;branch=0.
Via: SIP/2.0/UDP
192.168.0.83;rport=1063;received=68.80.200.100;branch=z9hG4bKe15e695c98bd9cb0.
From: "Paul (1002)"
<sip:9990010001@sip.mycompany.com;user=phone>;tag=4edd147cdac0025f.
To: <sip:14075551212@sip.mycompany.com;user=phone>;tag=0bf5212d.
Contact: <sip:9990010001@68.80.200.100:1063;user=phone>.
Proxy-Authorization: DIGEST username="9990010001",
realm="sip.mycompany.com", algorithm=MD5,
uri="sip:4075551212@216.229.127.60:5060",
nonce="419baee2139477c22db6d0ec9a47b23003512431",
response="7b83435f78bc60f5ced80304c7d54093".
Call-ID: aa31202374f54793(a)192.168.0.83.
CSeq: 12010 ACK.
User-Agent: Grandstream BT100 1.0.5.11.
Max-Forwards: 16.
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE.
Content-Length: 0.
.
#
I guess, you are doing record-route for ACK also. IMO, you should not
add record route headers for ACK.
I am not sure why Route header is completely removed from the request.
Proxy is supposed to delete only the top route value.
-Ranga
11:32 p.m.
You can safely add Record-Route header fields (this is what record_route
function does) to any SIP message. So just call record_route() without
the if clause.
User agents (including the registrar) would ignore Record-Route headers
in REGISTER messages anyway.
Your problem is related to Route header fields and these are not
generated by SER, but the user agents.
Jan.
On 17-11 23:13, Java Rockx wrote:
Ranga,
Yes, in my ser.cfg script I have
if (!method=="REGISTER") record-route();
I think there is plenty of confusion in the serusers list about wheather or not this is
correct.
Most of the example ser.cfg files I've seen, including those that sip with ser, show
this rather
than
if (!method=="INVITE") record-route();
Can anyone say which method is more RFC3261 compliant?
Also, I can't say wheather or not changing my ser.cfg to the latter will fix the
problems that our
PSTN provider has with our SIP dialogs. I'll find that out in the morning.
Regards,
Paul
--- Ranga <rangarao.v(a)gmail.com> wrote:
__________________________________
Do you Yahoo!?
The all-new My Yahoo! - Get yours free!
http://my.yahoo.com
U
2004/11/17 14:59:59.255342 68.80.200.100:1063 -> 68.80.201.101:5060
ACK sip:4075551212@216.229.127.60:5060 SIP/2.0.
Via: SIP/2.0/UDP 192.168.0.83;branch=z9hG4bKe15e695c98bd9cb0.
Route: <sip:68.80.201.101;ftag=4edd147cdac0025f;lr=on>.
Route: <sip:216.229.127.60:5060;lr>.
From: "Paul (1002)"
<sip:9990010001@sip.mycompany.com;user=phone>;tag=4edd147cdac0025f.
To: <sip:14075551212@sip.mycompany.com;user=phone>;tag=0bf5212d.
Contact: <sip:9990010001@192.168.0.83;user=phone>.
Proxy-Authorization: DIGEST username="9990010001",
realm="sip.mycompany.com", algorithm=MD5,
uri="sip:4075551212@216.229.127.60:5060",
nonce="419baee2139477c22db6d0ec9a47b23003512431",
response="7b83435f78bc60f5ced80304c7d54093".
Call-ID: aa31202374f54793(a)192.168.0.83.
CSeq: 12010 ACK.
User-Agent: Grandstream BT100 1.0.5.11.
Max-Forwards: 70.
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE.
Content-Length: 0.
.
#
U 2004/11/17 14:59:59.255737 68.80.201.101:5060 -> 216.229.127.60:5060
ACK sip:216.229.127.60:5060;lr SIP/2.0.
Record-Route: <sip:68.80.201.101;ftag=4edd147cdac0025f;lr=on>.
Via: SIP/2.0/UDP 68.80.201.101;branch=0.
Via: SIP/2.0/UDP
192.168.0.83;rport=1063;received=68.80.200.100;branch=z9hG4bKe15e695c98bd9cb0.
From: "Paul (1002)"
<sip:9990010001@sip.mycompany.com;user=phone>;tag=4edd147cdac0025f.
To: <sip:14075551212@sip.mycompany.com;user=phone>;tag=0bf5212d.
Contact: <sip:9990010001@68.80.200.100:1063;user=phone>.
Proxy-Authorization: DIGEST username="9990010001",
realm="sip.mycompany.com", algorithm=MD5,
uri="sip:4075551212@216.229.127.60:5060",
nonce="419baee2139477c22db6d0ec9a47b23003512431",
response="7b83435f78bc60f5ced80304c7d54093".
Call-ID: aa31202374f54793(a)192.168.0.83.
CSeq: 12010 ACK.
User-Agent: Grandstream BT100 1.0.5.11.
Max-Forwards: 16.
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE.
Content-Length: 0.
.
#
I guess, you are doing record-route for ACK also. IMO, you should not
add record route headers for ACK.
I am not sure why Route header is completely removed from the request.
Proxy is supposed to delete only the top route value.
-Ranga
7314
days inactive
7315
days old
6 comments
4 participants
participants (4)
-
Jan Janak -
Java Rockx -
Marian Dumitru -
Ranga