12 Oct
2006
12 Oct
'06
3:58 p.m.
Hi,
The external.proxy (not ser) rewrites the contact, as a way to be a
b2bua in terms of signalling: 10.111.0.20 is the external.proxy ...
10.111.0.50 is my.ser.proxy.
In terms of ACK processing, no doubt ser does its best ... it is just
the external.proxy not processing correctly the record-route, but this
shall be fixed soon ...
Meanwhile, i modify my config so i detect this kind of ACK and simply
t_relay them, nothing else.
Thanks everyone!
Cesc
On 10/12/06, Jan Janak <jan(a)iptel.org> wrote:
If you take a look at 200 OK you will notice that the
value of Contact
header is changed. The first 200 OK contains
sip:6007@10.111.0.144:5060
which I assume is correct. The second one contains:
sip:10.111.0.20:5060
But that (according to the dumps below) points to SER -- and that's the
problem. Do you rewrite Contact header in SER ? If so then it should not
point back to the SER instance.
SER treats the ACK correctly -- it has no Route headers and the
Request-URI points to SER -- hence it will process it as local message.
What you need to have fixed is the IP and port in rewritten contact.
Jan.
Cesc wrote:
Tks ... i guess i will try to modify the config
file so this ack
problem is taken care of in the most generic way i can think of ...
What i tried and it seems it didn't work is to enter the
"sip:10.111.0.20:5060" as a permanent entry in the location table
(using serctl). This way my config, which does a lookup(location) and
fails and drops the ACK, it would then return true and simply t_relay
it ...
as for the sjphone, i don't blame the phone. I think it does
reasonably good, given how bad the OK from the external.proxy gets to
him ...
Cesc
On 10/11/06, Greger V. Teigre <greger(a)teigre.com> wrote:
Short answer: yes.
Slightly longer: I have seen the same behavior. Without lr|lr=on the UA
will (correctly) go over to strict routing and use contact in r-uri
for the
ACK. As long as only your record-route disappeared (i.e. you are the only
hop in-between the UAS and UAC) this works. As your external proxy is
B2BUA
(at least for signalling), your fine if you relay the ACK.
I'm not sure exactly when this happens and why, I seem to remember I
posted
something on this behavior for sjphone a while back, but I'm not
capable of
finding it.
g-)
Cesc wrote:
Hello everyone!
I have my system based on ser 0.9.6. Internal calls work just fine.
I am now trying to interop with another system. It has a sort of
asterisk functionality, but it is not asterisk, it is a private
software company product.
my.phone ---------> my.ser.proxy -------------> external.proxy
-------------> external phone
10.111.0.119 10.111.0.50 10.111.0.20
10.111.0.144
There is no firewall or nat in the way.
The problem, i think, is that the external.proxy is buggy. I told
that to the company, but who knows when they will fix this.
* Look at the OK (message #9 and #10). My.ser.proxy record-routes all
invites. The Record-route headers reach the external.phone, which
copies them in the OK
message, sends the OK to the external.proxy ... and when this forwards
it to my.ser.proxy, they are gone! The OK reaches my.phone, but then
it generates my
problem: the ACK. It contains NO ROUTE headers and the r-uri is also
simply pointing to the external.proxy.
If the ACK had the ROUTE headers, my.ser.proxy would loose_route the
message and voila!
But as loose_route() returns false, my (maybe bad) config file gets
confused and treats it like a "new" call ... so it
lookup("location")
of the ACK r-ruri fails,
and the ACK is dropped.
Should i modify the config file so that ACK, if not loose_route'd, are
simply t_relay'd?
* They also modify the contact field. The reason is because behind the
external.proxy could be H323 or SIP phones, so they sort of want to be
a termination as far as signalling is concerned.
I attach the message flow, hope it helps ... Thanks!
Cesc
________________________________
No. Time Source Destination Protocol Info
1 0.000000000 10.111.0.119 10.111.0.50 SIP/SDP Request: INVITE
sip:6007@10.111.0.20, with session description
Session Initiation Protocol
Request-Line: INVITE sip:6007@10.111.0.20 SIP/2.0
Message Header
Via: SIP/2.0/UDP
10.111.0.119;rport;branch=z9hG4bK0a6f0077000004b2452bb819000009f00000191e
Content-Length: 337
Contact: <sip:7005@10.111.0.119:5060>
Call-ID: D95A2208-3031-44E8-862E-2877A042900D(a)10.111.0.119
Content-Type: application/sdp
CSeq: 1 INVITE
From: "7005"<sip:7005@10.111.0.50:5060>;tag=12083790462495
Max-Forwards: 70
To: <sip:6007@10.111.0.20>
User-Agent: SJphone/1.60.289a (SJ Labs)
Message body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): - 3369481881 3369481881 IN IP4
10.111.0.119
Session Name (s): SJphone
Connection Information (c): IN IP4 10.111.0.119
Time Description, active time (t): 0 0
Session Attribute (a): direction:active
Media Description, name and address (m): audio 49248 RTP/AVP 3 97 98 8 0
101
Media Attribute (a): rtpmap:3 GSM/8000
Media Attribute (a): rtpmap:97 iLBC/8000
Media Attribute (a): rtpmap:98 iLBC/8000
Media Attribute (a): fmtp:98 mode=20
Media Attribute (a): rtpmap:8 PCMA/8000
Media Attribute (a): rtpmap:0 PCMU/8000
Media Attribute (a): rtpmap:101 telephone-event/8000
Media Attribute (a): fmtp:101 0-11,16
No. Time Source Destination Protocol Info
3 0.001209000 10.111.0.50 10.111.0.20 SIP/SDP Request: INVITE
sip:6007@10.111.0.20:5060, with session description
Session Initiation Protocol
Request-Line: INVITE sip:6007@10.111.0.20:5060 SIP/2.0
Message Header
Record-Route: <sip:10.111.0.50;ftag=12083790462495;lr=on>
Via: SIP/2.0/UDP 10.111.0.50;branch=z9hG4bK2eff.75fbc652.0
Via: SIP/2.0/UDP
10.111.0.119;rport=5060;branch=z9hG4bK0a6f0077000004b2452bb819000009f00000191e
Content-Length: 337
Contact: <sip:7005@10.111.0.119:5060>
Call-ID: D95A2208-3031-44E8-862E-2877A042900D(a)10.111.0.119
Content-Type: application/sdp
CSeq: 1 INVITE
From: "7005"<sip:7005@10.111.0.50:5060>;tag=12083790462495
Max-Forwards: 16
To: <sip:6007@10.111.0.20>
User-Agent: SJphone/1.60.289a (SJ Labs)
Message body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): - 3369481881 3369481881 IN IP4
10.111.0.119
Session Name (s): SJphone
Connection Information (c): IN IP4 10.111.0.119
Time Description, active time (t): 0 0
Session Attribute (a): direction:active
Media Description, name and address (m): audio 49248 RTP/AVP 3 97 98 8 0
101
Media Attribute (a): rtpmap:3 GSM/8000
Media Attribute (a): rtpmap:97 iLBC/8000
Media Attribute (a): rtpmap:98 iLBC/8000
Media Attribute (a): fmtp:98 mode=20
Media Attribute (a): rtpmap:8 PCMA/8000
Media Attribute (a): rtpmap:0 PCMU/8000
Media Attribute (a): rtpmap:101 telephone-event/8000
Media Attribute (a): fmtp:101 0-11,16
No. Time Source Destination Protocol Info
5 0.040629000 10.111.0.20 10.111.0.144 SIP/SDP Request: INVITE
sip:6007@10.111.0.144:5060;transport=UDP, with session
description
Session Initiation Protocol
Request-Line: INVITE
sip:6007@10.111.0.144:5060;transport=UDP SIP/2.0
Message Header
Via: SIP/2.0/UDP 10.111.0.20:5060;branch=z9hG4bKm27749469
Via: SIP/2.0/UDP 10.111.0.50;branch=z9hG4bK2eff.75fbc652.0
Via: SIP/2.0/UDP
10.111.0.119;rport=5060;branch=z9hG4bK0a6f0077000004b2452bb819000009f00000191e
RECORD-ROUTE: <sip:10.111.0.50;ftag=12083790462495;lr=on>
From: "7005"<sip:7005@10.111.0.50:5060>;tag=12083790462495
To: <sip:6007@10.111.0.20>
Call-ID: D95A2208-3031-44E8-862E-2877A042900D(a)10.111.0.119
CSeq: 1 INVITE
Max-Forwards: 16
Contact: <sip:10.111.0.20>
User-Agent: SJphone/1.60.289a (SJ Labs)
Content-Type: application/sdp
Content-Length: 337
Message body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): - 3369481881 3369481881 IN IP4
10.111.0.119
Session Name (s): SJphone
Connection Information (c): IN IP4 10.111.0.119
Time Description, active time (t): 0 0
Session Attribute (a): direction:active
Media Description, name and address (m): audio 49248 RTP/AVP 3 97 98 8 0
101
Media Attribute (a): rtpmap:3 gsm/8000
Media Attribute (a): rtpmap:97 ilbc/8000
Media Attribute (a): rtpmap:98 ilbc/8000
Media Attribute (a): fmtp:98 mode=20
Media Attribute (a): rtpmap:8 pcma/8000
Media Attribute (a): rtpmap:0 pcmu/8000
Media Attribute (a): rtpmap:101 telephone-event/8000
Media Attribute (a): fmtp:101 0-11,16
No. Time Source Destination Protocol Info
9 3.736665000 10.111.0.144 10.111.0.20 SIP/SDP Status: 200 OK, with
session
description
Session Initiation Protocol
Status-Line: SIP/2.0 200 OK
Message Header
Via: SIP/2.0/UDP 10.111.0.20:5060;branch=z9hG4bKm27749469
Via: SIP/2.0/UDP 10.111.0.50;branch=z9hG4bK2eff.75fbc652.0
Via: SIP/2.0/UDP
10.111.0.119;rport=5060;branch=z9hG4bK0a6f0077000004b2452bb819000009f00000191e
Record-Route: <sip:10.111.0.50;ftag=12083790462495;lr=on>
Call-ID: D95A2208-3031-44E8-862E-2877A042900D(a)10.111.0.119
CSeq: 1 INVITE
From: "7005"<sip:7005@10.111.0.50:5060>;tag=12083790462495
To: <sip:6007@10.111.0.20>;tag=0rxFXxkiza46soF7
Contact: <sip:6007@10.111.0.144:5060>
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTER, REFER, NOTIFY, INFO,
PRACK, UPDATE
Supported: 100rel, replaces
Content-Type: application/sdp
Content-Length: 140
Message body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): 6007 14194531 23224216 IN IP4
10.111.0.144
Session Name (s): SIP CALL
Connection Information (c): IN IP4 10.111.0.144
Time Description, active time (t): 0 0
Media Description, name and address (m): audio 10000 RTP/AVP 8
Media Attribute (a): rtpmap:8 PCMA/8000
No. Time Source Destination Protocol Info
10 3.751306000 10.111.0.20 10.111.0.50 SIP/SDP Status: 200 OK, with
session
description
Session Initiation Protocol
Status-Line: SIP/2.0 200 OK
Message Header
Via: SIP/2.0/UDP 10.111.0.50;branch=z9hG4bK2eff.75fbc652.0
Via: SIP/2.0/UDP
10.111.0.119;rport=5060;branch=z9hG4bK0a6f0077000004b2452bb819000009f00000191e
From: "7005"<sip:7005@10.111.0.50:5060>;tag=12083790462495
To: <sip:6007@10.111.0.20>;tag=0rxFXxkiza46soF7
Call-ID: D95A2208-3031-44E8-862E-2877A042900D(a)10.111.0.119
CSeq: 1 INVITE
Contact: <sip:10.111.0.20:5060>
Content-Type: application/sdp
Content-Length: 140
Message body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): 6007 14194531 23224216 IN IP4
10.111.0.144
Session Name (s): SIP CALL
Connection Information (c): IN IP4 10.111.0.144
Time Description, active time (t): 0 0
Media Description, name and address (m): audio 10000 RTP/AVP 8
Media Attribute (a): rtpmap:8 pcma/8000
No. Time Source Destination Protocol Info
11 3.751589000 10.111.0.50 10.111.0.119 SIP/SDP Status: 200 OK, with
session description
Session Initiation Protocol
Status-Line: SIP/2.0 200 OK
Message Header
Via: SIP/2.0/UDP
10.111.0.119;rport=5060;branch=z9hG4bK0a6f0077000004b2452bb819000009f00000191e
From: "7005"<sip:7005@10.111.0.50:5060>;tag=12083790462495
To: <sip:6007@10.111.0.20>;tag=0rxFXxkiza46soF7
Call-ID: D95A2208-3031-44E8-862E-2877A042900D(a)10.111.0.119
CSeq: 1 INVITE
Contact: <sip:10.111.0.20:5060>
Content-Type: application/sdp
Content-Length: 140
Message body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): 6007 14194531 23224216 IN IP4
10.111.0.144
Session Name (s): SIP CALL
Connection Information (c): IN IP4 10.111.0.144
Time Description, active time (t): 0 0
Media Description, name and address (m): audio 10000 RTP/AVP 8
Media Attribute (a): rtpmap:8 pcma/8000
No. Time Source Destination Protocol Info
12 3.756127000 10.111.0.119 10.111.0.50 SIP Request: ACK
sip:10.111.0.20:5060
Session Initiation Protocol
Request-Line: ACK sip:10.111.0.20:5060 SIP/2.0
Message Header
Via: SIP/2.0/UDP
10.111.0.119;rport;branch=z9hG4bK0a6f0077000004b2452bb81d00000c5f00001922
Content-Length: 0
Call-ID: D95A2208-3031-44E8-862E-2877A042900D(a)10.111.0.119
CSeq: 1 ACK
From: "7005"<sip:7005@10.111.0.50:5060>;tag=12083790462495
Max-Forwards: 70
To: <sip:6007@10.111.0.20>;tag=0rxFXxkiza46soF7
User-Agent: SJphone/1.60.289a (SJ Labs)
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