21 Feb
2008
21 Feb
'08
9:40 p.m.
While it all works fine for our locally-registered users (and for some
of our peers), I'm still having problems from some peers dialing numbers
on our service. What happens is that we get an ACK (all our ACKs are
set to simply relay), and that ACK bounces around inside our server for
a while, then times out without ever replying -- which, after a short
period of time, cancels the connection for the peer, since they don't
see any ACK replies.
We start with a normal ACK sent from the remote peer (198.65.166.131) to
our server (22.33.44.55):
U 198.65.166.131:5060 -> 22.33.44.55:5060
ACK sip:301@22.33.44.55:5060 SIP/2.0.
Record-Route: <sip:198.65.166.131;ftag=j5j3wijb3t;lr>.
Via: SIP/2.0/UDP 198.65.166.131;branch=0.
Via: SIP/2.0/UDP
192.168.1.51:2051;received=63.95.254.25;branch=z9hG4bK-wv3vqqoh
qdm9;rport=2051.
Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
From: "User One"
<sip:17471111111@proxy01.sipphone.com:5060>;tag=j5j3wijb3t.
To: <sip:1101301@proxy01.sipphone.com;user=phone>;tag=as6c9ca1f6.
Call-ID: 3c26d1cab239-obtsvl9px3gg@snom190.
CSeq: 1 ACK.
Max-Forwards: 16.
Contact: <sip:17471111111@192.168.1.51:2051;line=ir1tukx9;nat=yes>.
Content-Length: 0.
P-hint: rr-enforced.
P-NATed-URI: YES (1).
P-RTP-Proxy: YES (1).
We end up getting things that look like this:
U 22.33.44.55:5060 -> 22.33.44.55:5060
ACK sip:301@22.33.44.55:5060 SIP/2.0.
Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
Record-Route: <sip:198.65.166.131;ftag=j5j3wijb3t;lr>.
Via: SIP/2.0/UDP 22.33.44.55;branch=0.
Via: SIP/2.0/UDP 22.33.44.55;branch=0.
Via: SIP/2.0/UDP 22.33.44.55;branch=0.
Via: SIP/2.0/UDP 22.33.44.55;branch=0.
Via: SIP/2.0/UDP 22.33.44.55;branch=0.
Via: SIP/2.0/UDP 22.33.44.55;branch=0.
Via: SIP/2.0/UDP 22.33.44.55;branch=0.
Via: SIP/2.0/UDP 22.33.44.55;branch=0.
Via: SIP/2.0/UDP 22.33.44.55;branch=0.
Via: SIP/2.0/UDP 22.33.44.55;branch=0.
Via: SIP/2.0/UDP 22.33.44.55;branch=0.
Via: SIP/2.0/UDP 22.33.44.55;branch=0.
Via: SIP/2.0/UDP 22.33.44.55;branch=0.
Via: SIP/2.0/UDP 22.33.44.55;branch=0.
Via: SIP/2.0/UDP 198.65.166.131;branch=0.
Via: SIP/2.0/UDP
192.168.1.51:2051;received=63.95.254.25;branch=z9hG4bK-wv3vqqoh
qdm9;rport=2051.
From: "User One"
<sip:17471111111@proxy01.sipphone.com:5060>;tag=j5j3wijb3t.
To: <sip:1101301@proxy01.sipphone.com;user=phone>;tag=as6c9ca1f6.
Call-ID: 3c26d1cab239-obtsvl9px3gg@snom190.
CSeq: 1 ACK.
Max-Forwards: 2.
Contact: <sip:17471111111@192.168.1.51:2051;line=ir1tukx9;nat=yes>.
Content-Length: 0.
P-hint: rr-enforced.
P-NATed-URI: YES (1).
P-RTP-Proxy: YES (1).
The top of that message shows that this is a message sent from our
server to our server, and it just tacks on a Record-Route and a Via line
one after another until it times out.
Again, my ACK handler is simply:
if(method=="ACK")
{
route(1);
break;
}
The record route is the very basic:
if(method!="REGISTER")
{
record_route();
};
And route[1] is just:
route[1]
{
if (!t_relay())
{
sl_reply_error();
};
}
All very basic stuff (and straight from the getting started pages I
think), and it mostly all works. Local users have no issues. Some peers
are okay.
But several of them are NOT, and I've tried everything I can think of
and can't get ACKs to behave. Can someone give me a bit of a clue as to
what to look at?
I don't understand the reason for tacking on another RR param each time
(nor do I REALLY understand why the lr=on is tacked on when this is
clearly not loose-routed). Some days, I feel like I really have a handle
on SER. Other days, I feel like I'm stumbling blindly. I never seem to
get to a level of comfort.
N.
21 Feb
21 Feb
9:54 p.m.
New subject: [Serusers] Still having problems with these ACKs (and BYEs).
It is not clear to me if the ACK in your script hits loose_route()....
It would be ideal if you could send the whole script.
Also the request-URI has the same IP address as the Route header field -- where
else shall the request go than to a loop?
-jiri
At 20:40 21/02/2008, SIP wrote:
While it all works fine for our locally-registered
users (and for some
of our peers), I'm still having problems from some peers dialing numbers
on our service. What happens is that we get an ACK (all our ACKs are
set to simply relay), and that ACK bounces around inside our server for
a while, then times out without ever replying -- which, after a short
period of time, cancels the connection for the peer, since they don't
see any ACK replies.
We start with a normal ACK sent from the remote peer (198.65.166.131) to
our server (22.33.44.55):
U 198.65.166.131:5060 -> 22.33.44.55:5060
ACK sip:301@22.33.44.55:5060 SIP/2.0.
Record-Route: <sip:198.65.166.131;ftag=j5j3wijb3t;lr>.
Via: SIP/2.0/UDP 198.65.166.131;branch=0.
Via: SIP/2.0/UDP
192.168.1.51:2051;received=63.95.254.25;branch=z9hG4bK-wv3vqqoh
qdm9;rport=2051.
Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
From: "User One"
<sip:17471111111@proxy01.sipphone.com:5060>;tag=j5j3wijb3t.
To: <sip:1101301@proxy01.sipphone.com;user=phone>;tag=as6c9ca1f6.
Call-ID: 3c26d1cab239-obtsvl9px3gg@snom190.
CSeq: 1 ACK.
Max-Forwards: 16.
Contact: <sip:17471111111@192.168.1.51:2051;line=ir1tukx9;nat=yes>.
Content-Length: 0.
P-hint: rr-enforced.
P-NATed-URI: YES (1).
P-RTP-Proxy: YES (1).
We end up getting things that look like this:
U 22.33.44.55:5060 -> 22.33.44.55:5060
ACK sip:301@22.33.44.55:5060 SIP/2.0.
Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
Record-Route: <sip:198.65.166.131;ftag=j5j3wijb3t;lr>.
Via: SIP/2.0/UDP 22.33.44.55;branch=0.
Via: SIP/2.0/UDP 22.33.44.55;branch=0.
Via: SIP/2.0/UDP 22.33.44.55;branch=0.
Via: SIP/2.0/UDP 22.33.44.55;branch=0.
Via: SIP/2.0/UDP 22.33.44.55;branch=0.
Via: SIP/2.0/UDP 22.33.44.55;branch=0.
Via: SIP/2.0/UDP 22.33.44.55;branch=0.
Via: SIP/2.0/UDP 22.33.44.55;branch=0.
Via: SIP/2.0/UDP 22.33.44.55;branch=0.
Via: SIP/2.0/UDP 22.33.44.55;branch=0.
Via: SIP/2.0/UDP 22.33.44.55;branch=0.
Via: SIP/2.0/UDP 22.33.44.55;branch=0.
Via: SIP/2.0/UDP 22.33.44.55;branch=0.
Via: SIP/2.0/UDP 22.33.44.55;branch=0.
Via: SIP/2.0/UDP 198.65.166.131;branch=0.
Via: SIP/2.0/UDP
192.168.1.51:2051;received=63.95.254.25;branch=z9hG4bK-wv3vqqoh
qdm9;rport=2051.
From: "User One"
<sip:17471111111@proxy01.sipphone.com:5060>;tag=j5j3wijb3t.
To: <sip:1101301@proxy01.sipphone.com;user=phone>;tag=as6c9ca1f6.
Call-ID: 3c26d1cab239-obtsvl9px3gg@snom190.
CSeq: 1 ACK.
Max-Forwards: 2.
Contact: <sip:17471111111@192.168.1.51:2051;line=ir1tukx9;nat=yes>.
Content-Length: 0.
P-hint: rr-enforced.
P-NATed-URI: YES (1).
P-RTP-Proxy: YES (1).
The top of that message shows that this is a message sent from our
server to our server, and it just tacks on a Record-Route and a Via line
one after another until it times out.
Again, my ACK handler is simply:
if(method=="ACK")
{
route(1);
break;
}
The record route is the very basic:
if(method!="REGISTER")
{
record_route();
};
And route[1] is just:
route[1]
{
if (!t_relay())
{
sl_reply_error();
};
}
All very basic stuff (and straight from the getting started pages I
think), and it mostly all works. Local users have no issues. Some peers
are okay.
But several of them are NOT, and I've tried everything I can think of
and can't get ACKs to behave. Can someone give me a bit of a clue as to
what to look at?
I don't understand the reason for tacking on another RR param each time
(nor do I REALLY understand why the lr=on is tacked on when this is
clearly not loose-routed). Some days, I feel like I really have a handle
on SER. Other days, I feel like I'm stumbling blindly. I never seem to
get to a level of comfort.
N.
_______________________________________________
Serusers mailing list
Serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
--
Jiri Kuthan http://iptel.org/~jiri/
11:02 p.m.
New subject: [Serusers] Still having problems with these ACKs (and BYEs).
According to the log statement I placed into the loose_route() section,
this ACK never hits loose_route at any point.
And yeah... the whole ACK thing is confusing me. Not sure where it's
supposed to go in there.
By sending you the whole script, what do you mean? Do you want me to
send you the whole config (I have to warn you, it's 51k and rather
complex) or the whole log? Or both?
N.
Jiri Kuthan wrote:
It is not clear to me if the ACK in your script hits
loose_route()....
It would be ideal if you could send the whole script.
Also the request-URI has the same IP address as the Route header field -- where
else shall the request go than to a loop?
-jiri
At 20:40 21/02/2008, SIP wrote:
While it all works fine for our
locally-registered users (and for some
of our peers), I'm still having problems from some peers dialing numbers
on our service. What happens is that we get an ACK (all our ACKs are
set to simply relay), and that ACK bounces around inside our server for
a while, then times out without ever replying -- which, after a short
period of time, cancels the connection for the peer, since they don't
see any ACK replies.
We start with a normal ACK sent from the remote peer (198.65.166.131) to
our server (22.33.44.55):
U 198.65.166.131:5060 -> 22.33.44.55:5060
ACK sip:301@22.33.44.55:5060 SIP/2.0.
Record-Route: <sip:198.65.166.131;ftag=j5j3wijb3t;lr>.
Via: SIP/2.0/UDP 198.65.166.131;branch=0.
Via: SIP/2.0/UDP
192.168.1.51:2051;received=63.95.254.25;branch=z9hG4bK-wv3vqqoh
qdm9;rport=2051.
Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
From: "User One"
<sip:17471111111@proxy01.sipphone.com:5060>;tag=j5j3wijb3t.
To: <sip:1101301@proxy01.sipphone.com;user=phone>;tag=as6c9ca1f6.
Call-ID: 3c26d1cab239-obtsvl9px3gg@snom190.
CSeq: 1 ACK.
Max-Forwards: 16.
Contact: <sip:17471111111@192.168.1.51:2051;line=ir1tukx9;nat=yes>.
Content-Length: 0.
P-hint: rr-enforced.
P-NATed-URI: YES (1).
P-RTP-Proxy: YES (1).
We end up getting things that look like this:
U 22.33.44.55:5060 -> 22.33.44.55:5060
ACK sip:301@22.33.44.55:5060 SIP/2.0.
Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
Record-Route: <sip:198.65.166.131;ftag=j5j3wijb3t;lr>.
Via: SIP/2.0/UDP 22.33.44.55;branch=0.
Via: SIP/2.0/UDP 22.33.44.55;branch=0.
Via: SIP/2.0/UDP 22.33.44.55;branch=0.
Via: SIP/2.0/UDP 22.33.44.55;branch=0.
Via: SIP/2.0/UDP 22.33.44.55;branch=0.
Via: SIP/2.0/UDP 22.33.44.55;branch=0.
Via: SIP/2.0/UDP 22.33.44.55;branch=0.
Via: SIP/2.0/UDP 22.33.44.55;branch=0.
Via: SIP/2.0/UDP 22.33.44.55;branch=0.
Via: SIP/2.0/UDP 22.33.44.55;branch=0.
Via: SIP/2.0/UDP 22.33.44.55;branch=0.
Via: SIP/2.0/UDP 22.33.44.55;branch=0.
Via: SIP/2.0/UDP 22.33.44.55;branch=0.
Via: SIP/2.0/UDP 22.33.44.55;branch=0.
Via: SIP/2.0/UDP 198.65.166.131;branch=0.
Via: SIP/2.0/UDP
192.168.1.51:2051;received=63.95.254.25;branch=z9hG4bK-wv3vqqoh
qdm9;rport=2051.
From: "User One"
<sip:17471111111@proxy01.sipphone.com:5060>;tag=j5j3wijb3t.
To: <sip:1101301@proxy01.sipphone.com;user=phone>;tag=as6c9ca1f6.
Call-ID: 3c26d1cab239-obtsvl9px3gg@snom190.
CSeq: 1 ACK.
Max-Forwards: 2.
Contact: <sip:17471111111@192.168.1.51:2051;line=ir1tukx9;nat=yes>.
Content-Length: 0.
P-hint: rr-enforced.
P-NATed-URI: YES (1).
P-RTP-Proxy: YES (1).
The top of that message shows that this is a message sent from our
server to our server, and it just tacks on a Record-Route and a Via line
one after another until it times out.
Again, my ACK handler is simply:
if(method=="ACK")
{
route(1);
break;
}
The record route is the very basic:
if(method!="REGISTER")
{
record_route();
};
And route[1] is just:
route[1]
{
if (!t_relay())
{
sl_reply_error();
};
}
All very basic stuff (and straight from the getting started pages I
think), and it mostly all works. Local users have no issues. Some peers
are okay.
But several of them are NOT, and I've tried everything I can think of
and can't get ACKs to behave. Can someone give me a bit of a clue as to
what to look at?
I don't understand the reason for tacking on another RR param each time
(nor do I REALLY understand why the lr=on is tacked on when this is
clearly not loose-routed). Some days, I feel like I really have a handle
on SER. Other days, I feel like I'm stumbling blindly. I never seem to
get to a level of comfort.
N.
_______________________________________________
Serusers mailing list
Serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
--
Jiri Kuthan http://iptel.org/~jiri/
11:29 p.m.
New subject: [Serusers] Still having problems with these ACKs (and BYEs).
1) I would need the real ACK (yours seems to be changed for privacy reasons?)--I'm
a bit surprised to see same IP in rURI and Route; that can't really work;
I'm guessing that was a rewriting mistake?
2) I would need the config file (with some disclaimers -- not sure when and
how fast I can parse 51k :-))
-jiri
At 22:02 21/02/2008, SIP wrote:
According to the log statement I placed into the
loose_route() section,
this ACK never hits loose_route at any point.
And yeah... the whole ACK thing is confusing me. Not sure where it's
supposed to go in there.
By sending you the whole script, what do you mean? Do you want me to
send you the whole config (I have to warn you, it's 51k and rather
complex) or the whole log? Or both?
N.
Jiri Kuthan wrote:
--
Jiri Kuthan http://iptel.org/~jiri/
It is not clear to me if the ACK in your script
hits loose_route()....
It would be ideal if you could send the whole script.
Also the request-URI has the same IP address as the Route header field -- where
else shall the request go than to a loop?
-jiri
At 20:40 21/02/2008, SIP wrote:
_______________________________________________
Serusers mailing list
Serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers While it all works fine for our
locally-registered users (and for some
of our peers), I'm still having problems from some peers dialing numbers
on our service. What happens is that we get an ACK (all our ACKs are
set to simply relay), and that ACK bounces around inside our server for
a while, then times out without ever replying -- which, after a short
period of time, cancels the connection for the peer, since they don't
see any ACK replies.
We start with a normal ACK sent from the remote peer (198.65.166.131) to
our server (22.33.44.55):
U 198.65.166.131:5060 -> 22.33.44.55:5060
ACK sip:301@22.33.44.55:5060 SIP/2.0.
Record-Route: <sip:198.65.166.131;ftag=j5j3wijb3t;lr>.
Via: SIP/2.0/UDP 198.65.166.131;branch=0.
Via: SIP/2.0/UDP
192.168.1.51:2051;received=63.95.254.25;branch=z9hG4bK-wv3vqqoh
qdm9;rport=2051.
Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
From: "User One"
<sip:17471111111@proxy01.sipphone.com:5060>;tag=j5j3wijb3t.
To: <sip:1101301@proxy01.sipphone.com;user=phone>;tag=as6c9ca1f6.
Call-ID: 3c26d1cab239-obtsvl9px3gg@snom190.
CSeq: 1 ACK.
Max-Forwards: 16.
Contact: <sip:17471111111@192.168.1.51:2051;line=ir1tukx9;nat=yes>.
Content-Length: 0.
P-hint: rr-enforced.
P-NATed-URI: YES (1).
P-RTP-Proxy: YES (1).
We end up getting things that look like this:
U 22.33.44.55:5060 -> 22.33.44.55:5060
ACK sip:301@22.33.44.55:5060 SIP/2.0.
Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
Record-Route: <sip:22.33.44.55;ftag=j5j3wijb3t;lr=on>.
Record-Route: <sip:198.65.166.131;ftag=j5j3wijb3t;lr>.
Via: SIP/2.0/UDP 22.33.44.55;branch=0.
Via: SIP/2.0/UDP 22.33.44.55;branch=0.
Via: SIP/2.0/UDP 22.33.44.55;branch=0.
Via: SIP/2.0/UDP 22.33.44.55;branch=0.
Via: SIP/2.0/UDP 22.33.44.55;branch=0.
Via: SIP/2.0/UDP 22.33.44.55;branch=0.
Via: SIP/2.0/UDP 22.33.44.55;branch=0.
Via: SIP/2.0/UDP 22.33.44.55;branch=0.
Via: SIP/2.0/UDP 22.33.44.55;branch=0.
Via: SIP/2.0/UDP 22.33.44.55;branch=0.
Via: SIP/2.0/UDP 22.33.44.55;branch=0.
Via: SIP/2.0/UDP 22.33.44.55;branch=0.
Via: SIP/2.0/UDP 22.33.44.55;branch=0.
Via: SIP/2.0/UDP 22.33.44.55;branch=0.
Via: SIP/2.0/UDP 198.65.166.131;branch=0.
Via: SIP/2.0/UDP
192.168.1.51:2051;received=63.95.254.25;branch=z9hG4bK-wv3vqqoh
qdm9;rport=2051.
From: "User One"
<sip:17471111111@proxy01.sipphone.com:5060>;tag=j5j3wijb3t.
To: <sip:1101301@proxy01.sipphone.com;user=phone>;tag=as6c9ca1f6.
Call-ID: 3c26d1cab239-obtsvl9px3gg@snom190.
CSeq: 1 ACK.
Max-Forwards: 2.
Contact: <sip:17471111111@192.168.1.51:2051;line=ir1tukx9;nat=yes>.
Content-Length: 0.
P-hint: rr-enforced.
P-NATed-URI: YES (1).
P-RTP-Proxy: YES (1).
The top of that message shows that this is a message sent from our
server to our server, and it just tacks on a Record-Route and a Via line
one after another until it times out.
Again, my ACK handler is simply:
if(method=="ACK")
{
route(1);
break;
}
The record route is the very basic:
if(method!="REGISTER")
{
record_route();
};
And route[1] is just:
route[1]
{
if (!t_relay())
{
sl_reply_error();
};
}
All very basic stuff (and straight from the getting started pages I
think), and it mostly all works. Local users have no issues. Some peers
are okay.
But several of them are NOT, and I've tried everything I can think of
and can't get ACKs to behave. Can someone give me a bit of a clue as to
what to look at?
I don't understand the reason for tacking on another RR param each time
(nor do I REALLY understand why the lr=on is tacked on when this is
clearly not loose-routed). Some days, I feel like I really have a handle
on SER. Other days, I feel like I'm stumbling blindly. I never seem to
get to a level of comfort.
N.
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Jiri Kuthan http://iptel.org/~jiri/
6125
days inactive
6125
days old
3 comments
2 participants
participants (2)
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Jiri Kuthan -
SIP