[Serusers] Re: [Users] Route, Record Route and quintum
Rick Thompson
rthompson at vir2com.com
Fri Sep 9 21:33:09 CEST 2005
Ouch!!
And I thought I had problems with Quintum!!! I've spent what seems like a
life time trying to get Quintum Tenors to work with ser and nated ua's. For
the life of me I can't figure out why a ua behind a Linksys nat router works
with fix_nated_sdp but it doesn't with force_rtp_proxy. The setup is ua ==>
nat ==> ser ==> quintum ==> PSTN. And a ua behind a Dlink nat router works
with force_rtp_proxy but doesn't work with fix_nated_sdp. But if I use
asterisk as a pstn gw, everything works fine with either sdp or rtp_proxy.
Maybe if I could detect when each occurred I could implement the working
configuration. Does anyone have any experience with this type of
configuration? (systemic verses cone nat) I sure could use the help before
I lose all my hair. Thanx
_____
From: serusers-bounces at iptel.org [mailto:serusers-bounces at iptel.org] On
Behalf Of Kanakatti M. Subramanya
Sent: Thursday, September 08, 2005 10:53 PM
Cc: serusers at iptel.org; users openser.org
Subject: Re: [Serusers] Re: [Users] Route, Record Route and quintum
Brilliantly, and quite eloquently detailed Jan. Thanx!
FYI, I spent a *looong* time flailing around with Quintium hardware trying
to get it to do the *correct* thing (and it was for a three person account
too!)
Finally gave up in disgust, depression and despair.
Mind you, this was just before I discovered that one of our major providers
was switching to Sonus.
Sonus, I ask you.
Yikes....
Jan Janak wrote:
OK, I think I know where the confusion comes from, it is like this:
When a user agent sends a SIP request, the request will traverse one ore
more SIP proxies until it reaches the target user agent. Each proxy and
user agent along the path of the request will add its Via header field.
The purpose of Via header fields is to make sure that all replies will
traverse exactly the same set of proxies (but in reverse order) as the
request. In other words Via header fields are used to route _replies_.
They record the path of the request so that replies can follow it.
Route and Record-Route header fields have a slightly different purpose
-- they are used to route _requests_. Let's assume this scenario:
INV INV
UA A --------> proxy ----------> UA B
a at 1.2.3.4 b at 5.6.7.8
User agent A with IP address 1.2.3.4 sends an INVITE request to the
proxy which forwards the request to user agent B. UA A adds the Contact
header field in the request. The Contact header field tells UA B that it
can reach UA A on IP 1.2.3.4. UA B adds a Contact header field to 200 OK
reply, telling UA A that it can reach B on IP address 5.6.7.8.
This way both user agents exchange their IP addresses and they do not
need the proxy anymore. They can easily send all further SIP messages
directly to each other, because they remember the IP address of the
remote party from Contact header field. This way all further requests
would bypass the proxy server.
There are many cases where the proxy server needs to see _all_ future
SIP messages exchanged between the user agent (for example when
performing accounting). In this case the proxy server needs tell the
user agents that they should not exchange future SIP messages directly,
but they should relay them through the proxy again.
The proxy server can do this by inserting Record-Route header field in
the INVITE message. The Record-Route header field contains the IP
address of the proxy and once the INVITE message reaches UA B, it extracts
the IP address of the proxy server from Record-Route header field and
store it in memory along with the IP address of the remote party (UA A).
So UA B knows the IP address of UA A and it also knows that it should
send all future requests to UA A through the proxy server.
UA A should also send all future SIP requests to UA B through the proxy
server, but it does not know it yet, because the proxy server added
Record-Route header field to the INVITE message (which will only reach
UA B). UA B has to tell UA A that all future SIP requests should be sent
through the proxy, and it does so by copying all Record-Route header
fields (in our example there will be only one) from the INVITE message to
200 OK (which is sent from UA B to UA A). UA A will then extract
Record-Route header fields along with Contact from 200 OK and store it
in memory. At this point both user agents know the IP address of the
remote party and that they should relay all SIP messages through the
proxy.
Now what happens when UA A wants to send an ACK to UA B. It will lookup
the IP address of the remote party (Contact from 200 OK) from memory and put
it in the Request-URI of ACK. The reason why ACK does not have the same
Request-URI as the original INVITE is that the ACK should be sent to the
user agent instance that generated 200 OK -- the Request-URI from the
INVITE would fork if user B had several user agents and this is not
desirable for ACKs. The URI based on Contact header field never forks,
it is delivered only to the UA instance that generated 200 OK.
UA A will also find the URI from the Record-Route header field in memory
(stored along with the contact of the remote party). It will create a
Route header field, put the URI in it and append it to the ACK.
RFC3261 says that if there is a Route header field in a SIP message,
then the message should not be sent to the URI in the Request-URI, but
to the URI in Route header field and thus the ACK would be sent to the
proxy. The proxy will then remove the Route header field and because
there is no other Route header field in our message, it will forward the
request to the Request-URI which will take the message to UA B.
Note the difference between Record-Route and Route header fields.
Record-Route header fields tell SIP user agent and proxies where
_future_ SIP requests should be sent. Route header field are constructed
from Record-Route header fields and they tell user agent and proxies
where the request that contains the Route header fields should be sent.
Two header field names are needed to avoid confusion, because a proxy
server can add Record-Route header field to _any_ SIP request, including
requests that already contain Route header fields (using the same name
would mix them).
The relationship between the two user agents (when they remember the IP
address of the remote party and IPs of proxies) is called dialog. SIP
requests that are forwarded using the information extracted from
Contacts and Record-Route header fields are called "SIP requests within
dialog" (typically ACK and BYE).
When a user agent sends a SIP request within a dialog, it will always
put the Contact of the remote party in the Request-URI and copy all URIs
extracted from Record-Route header fields in Route header fields (in
either forward or reverse order, based on the direction):
ACK sip:b at 5.6.7.8 SIP/2.0
Route: sip:proxy at proxy.ip;lr
...
This is how all recent SIP implementations should behave. Always put the
Contact of the remote party into the Request-URI and list all proxies that
inserted Record-Route in Route header fields appended to the message.
Plain and simple.
Unfortunately there are also older SIP implementations (from pre-RFC3261
era) that need special care, because record-routing worked differently
then. In this case the Record-Route does not contain the contact of
the remote party but the topmost Route header field, and the value of
the remote contact is preserved in the last Route header field in the
message. This "compatibility mode" is what makes record routing complex
and hard to understand, but you can ignore it (you can always find
details in RFC3261).
In conclusion:
1) Via is used to route responses
2) Route and Record-Route headers are used to route requests
3) Proxies use Record-Route to signal that they want to stay on the path
of future requests.
4) User agents use Record-Route headers to build Route headers.
5) Contact of the remote party is always put in the Request-URI of
requests in dialogs.
6) The list of Route header field is created from the list of
Record-Route header fields.
7) The URI in the topmost Route header field overrides the URI in the
Request-URI, so the Request-URI will only be used if there are no
more Route header fields.
Jan.
PS: When talking to Quintum guys, you should ask them to implement what
I just described:
1) Take the Contact header field from 200 OK and put it in the
Request-URI of ACK.
2) Take all Record-Route header fields from 200 OK, reverse their
order, and append them as Route header fields to the ACK.
3) Send the ACK to the topmost Route, if any, otherwise send to the
Request-URI.
On 08-09-2005 19:23, Iqbal wrote:
Hi
Just in case I missed something, and before I trash quintum, I have put
the entire dialogue below, to make sure I am getting it correct. If
someone can take a quick look, cause loose routes and strict are driving
me crazy
The setup is quintum --- ser---gw (a.b.c.d ----x.x.x.x ----z.z.z.z)
quintum IP = a.b.c.d
ser ip = x.x.x.x
gw = z.z.z.z
gw2 = y.y.y.y
Now you cannot get to gw2 direct
I have read the RFC, and it may as well be written in French. Also the
Via headers which messages follow those.
>From what i see we have
the RURI, The route headers and the via headers, which is followed and when
iqbal
U 2005/09/06 15:23:06.414932 a.b.c.d:5060 -> x.x.x.x:5060
INVITE sip:07866318220 at x.x.x.x SIP/2.0.
CSeq: 1 INVITE.
Call-ID: call-00CC9E9D-4FBE-D311-0207-A at a.b.c.d.
Contact: <mailto:sip:046677 at a.b.c.d> <sip:046677 at a.b.c.d>.
Content-Type: application/sdp.
From: <mailto:sip:046677 at a.b.c.d> <sip:046677 at a.b.c.d>;tag=c3ac7b24-a.
Session-GUID: 859321956-876164920-926430820-875782963.
To: <mailto:sip:123456789 at x.x.x.x> <sip:123456789 at x.x.x.x>.
Via: SIP/2.0/UDP a.b.c.d;branch=z9hG4bK-tenor-c3ac-7b24-0018.
Content-Length: 229.
User-Agent: Quintum/1.0.0.
Quintum:
0c01030b0232360501000715000000000000000d006c0a828180803034363637371301000f0b
413031322d313033353941.
Max-Forwards: 70.
.
v=0.
o=Quintum 7 7 IN IP4 a.b.c.d.
s=VoipCall.
c=IN IP4 a.b.c.d.
t=0 0.
m=audio 10254 RTP/AVP 18 4 101.
c=IN IP4 a.b.c.d.
a=rtpmap:18 g729/8000/1.
a=rtpmap:4 g723/8000/1.
a=rtpmap:101 telephone-event/8000/1.
#
U 2005/09/06 15:23:06.452476 x.x.x.x:5060 -> a.b.c.d:5060
SIP/2.0 100 trying -- your call is important to us.
CSeq: 1 INVITE.
Call-ID: call-00CC9E9D-4FBE-D311-0207-A at a.b.c.d.
From: <mailto:sip:046677 at a.b.c.d> <sip:046677 at a.b.c.d>;tag=c3ac7b24-a.
To: <mailto:sip:123456789 at x.x.x.x> <sip:123456789 at x.x.x.x>.
Via: SIP/2.0/UDP a.b.c.d;branch=z9hG4bK-tenor-c3ac-7b24-0018.
Content-Length: 0.
Warning: 392 x.x.x.x:5060 "Noisy feedback tells: pid=24439
req_src_ip=a.b.c.d req_src_port=5060 in_uri=sip:123456789 at x.x.x.x
out_uri=sip:447866318220 at 213.166.5.135:5060 via_cnt==1".
.
#
U 2005/09/06 15:23:06.453034 x.x.x.x:5060 -> z.z.z.z:5060
INVITE sip:44123456789 at z.z.z.z:5060 SIP/2.0.
Record-Route: <sip:x.x.x.x;ftag=c3ac7b24-a;lr=on>.
CSeq: 1 INVITE.
Call-ID: call-00CC9E9D-4FBE-D311-0207-A at a.b.c.d.
Contact: <mailto:sip:046677 at a.b.c.d> <sip:046677 at a.b.c.d>.
Content-Type: application/sdp.
From: <mailto:sip:046677 at a.b.c.d> <sip:046677 at a.b.c.d>;tag=c3ac7b24-a.
Session-GUID: 859321956-876164920-926430820-875782963.
To: <mailto:sip:123456789 at x.x.x.x> <sip:123456789 at x.x.x.x>.
Via: SIP/2.0/UDP x.x.x.x;branch=z9hG4bKb322.930d8117.0.
Via: SIP/2.0/UDP a.b.c.d;branch=z9hG4bK-tenor-c3ac-7b24-0018.
Content-Length: 229.
User-Agent: Quintum/1.0.0.
Quintum:
0c01030b0232360501000715000000000000000d006c0a828180803034363637371301000f0b
413031322d313033353941.
Max-Forwards: 16.
.
v=0.
o=Quintum 7 7 IN IP4 a.b.c.d.
s=VoipCall.
c=IN IP4 a.b.c.d.
t=0 0.
m=audio 10254 RTP/AVP 18 4 101.
c=IN IP4 a.b.c.d.
a=rtpmap:18 g729/8000/1.
a=rtpmap:4 g723/8000/1.
a=rtpmap:101 telephone-event/8000/1.
#
U 2005/09/06 15:23:06.465531 z.z.z.z:5060 -> x.x.x.x:5060
SIP/2.0 100 trying -- your call is important to us.
CSeq: 1 INVITE.
Call-ID: call-00CC9E9D-4FBE-D311-0207-A at a.b.c.d.
From: <mailto:sip:046677 at a.b.c.d> <sip:046677 at a.b.c.d>;tag=c3ac7b24-a.
To: <mailto:sip:123456789 at x.x.x.x> <sip:123456789 at x.x.x.x>.
Via: SIP/2.0/UDP x.x.x.x;branch=z9hG4bKb322.930d8117.0.
Via: SIP/2.0/UDP a.b.c.d;branch=z9hG4bK-tenor-c3ac-7b24-0018.
Server: Sip EXpress router (0.8.99-dev (i386/linux)).
Content-Length: 0.
.
########
U 2005/09/06 15:23:12.785564 z.z.z.z:5060 -> x.x.x.x:5060
SIP/2.0 183 Session Progress.
Via: SIP/2.0/UDP x.x.x.x;branch=z9hG4bKb322.930d8117.0,SIP/2.0/UDP
a.b.c.d;branch=z9hG4bK-tenor-c3ac-7b24-0018.
From: <mailto:sip:046677 at a.b.c.d> <sip:046677 at a.b.c.d>;tag=c3ac7b24-a.
To: <mailto:sip:123456789 at x.x.x.x>
<sip:123456789 at x.x.x.x>;tag=1A4297F4-2C9.
Date: Tue, 06 Sep 2005 14:07:59 gmt.
Call-ID: call-00CC9E9D-4FBE-D311-0207-A at a.b.c.d.
Server: Cisco-SIPGateway/IOS-12.x.
CSeq: 1 INVITE.
Allow-Events: telephone-event.
Contact: <mailto:sip:0044123456789 at y.y.y.y:5060>
<sip:0044123456789 at y.y.y.y:5060>.
Record-Route:
<sip:z.z.z.z;ftag=c3ac7b24-a;lr=on>,<sip:x.x.x.x;ftag=c3ac7b24-a;lr=on>.
Content-Disposition: session;handling=required.
Content-Type: application/sdp.
Content-Length: 206.
.
v=0.
o=CiscoSystemsSIP-GW-UserAgent 7438 1091 IN IP4 y.y.y.y.
s=SIP Call.
c=IN IP4 y.y.y.y.
t=0 0.
m=audio 18590 RTP/AVP 18.
c=IN IP4 y.y.y.y.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=yes.
-------------------
This is where the confusion starts, what path should the 183 take, or
how is the path decided, is it the RURI, or from the RR, or does it not
matter for 183
#
U 2005/09/06 15:23:12.788274 x.x.x.x:5060 -> a.b.c.d:5060
SIP/2.0 183 Session Progress.
Via: SIP/2.0/UDP a.b.c.d;branch=z9hG4bK-tenor-c3ac-7b24-0018.
From: <mailto:sip:046677 at a.b.c.d> <sip:046677 at a.b.c.d>;tag=c3ac7b24-a.
To: <mailto:sip:123456789 at x.x.x.x>
<sip:123456789 at x.x.x.x>;tag=1A4297F4-2C9.
Date: Tue, 06 Sep 2005 14:07:59 gmt.
Call-ID: call-00CC9E9D-4FBE-D311-0207-A at a.b.c.d.
Server: Cisco-SIPGateway/IOS-12.x.
CSeq: 1 INVITE.
Allow-Events: telephone-event.
Contact: <mailto:sip:0044123456789 at y.y.y.y:5060>
<sip:0044123456789 at y.y.y.y:5060>.
Record-Route:
<sip:z.z.z.z;ftag=c3ac7b24-a;lr=on>,<sip:x.x.x.x;ftag=c3ac7b24-a;lr=on>.
Content-Disposition: session;handling=required.
Content-Type: application/sdp.
Content-Length: 206.
.
v=0.
o=CiscoSystemsSIP-GW-UserAgent 7438 1091 IN IP4 y.y.y.y.
s=SIP Call.
c=IN IP4 y.y.y.y.
t=0 0.
m=audio 18590 RTP/AVP 18.
c=IN IP4 y.y.y.y.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=yes.
----------------------
and the OK below, should it not have gone to zzzz and then xxxx, as in
RR, and should not the matching RR be stripped off
----------------------
###############
U 2005/09/06 15:23:16.610600 z.z.z.z:5060 -> x.x.x.x:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP x.x.x.x;branch=z9hG4bKb322.930d8117.0,SIP/2.0/UDP
a.b.c.d;branch=z9hG4bK-tenor-c3ac-7b24-0018.
From: <mailto:sip:046677 at a.b.c.d> <sip:046677 at a.b.c.d>;tag=c3ac7b24-a.
To: <mailto:sip:123456789 at x.x.x.x>
<sip:123456789 at x.x.x.x>;tag=1A4297F4-2C9.
Date: Tue, 06 Sep 2005 14:07:59 gmt.
Call-ID: call-00CC9E9D-4FBE-D311-0207-A at a.b.c.d.
Server: Cisco-SIPGateway/IOS-12.x.
CSeq: 1 INVITE.
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER,
SUBSCRIBE, NOTIFY, INFO.
Allow-Events: telephone-event.
Contact: <mailto:sip:0044123456789 at y.y.y.y:5060>
<sip:0044123456789 at y.y.y.y:5060>.
Record-Route:
<sip:z.z.z.z;ftag=c3ac7b24-a;lr=on>,<sip:x.x.x.x;ftag=c3ac7b24-a;lr=on>.
Content-Type: application/sdp.
Content-Length: 206.
.
v=0.
o=CiscoSystemsSIP-GW-UserAgent 7438 1091 IN IP4 y.y.y.y.
s=SIP Call.
c=IN IP4 y.y.y.y.
t=0 0.
m=audio 18590 RTP/AVP 18.
c=IN IP4 y.y.y.y.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=yes.
#
U 2005/09/06 15:23:16.613003 x.x.x.x:5060 -> a.b.c.d:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP a.b.c.d;branch=z9hG4bK-tenor-c3ac-7b24-0018.
From: <mailto:sip:046677 at a.b.c.d> <sip:046677 at a.b.c.d>;tag=c3ac7b24-a.
To: <mailto:sip:123456789 at x.x.x.x>
<sip:123456789 at x.x.x.x>;tag=1A4297F4-2C9.
Date: Tue, 06 Sep 2005 14:07:59 gmt.
Call-ID: call-00CC9E9D-4FBE-D311-0207-A at a.b.c.d.
Server: Cisco-SIPGateway/IOS-12.x.
CSeq: 1 INVITE.
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER,
SUBSCRIBE, NOTIFY, INFO.
Allow-Events: telephone-event.
Contact: <mailto:sip:0044123456789 at y.y.y.y:5060>
<sip:0044123456789 at y.y.y.y:5060>.
Record-Route:
<sip:z.z.z.z;ftag=c3ac7b24-a;lr=on>,<sip:x.x.x.x;ftag=c3ac7b24-a;lr=on>.
Content-Type: application/sdp.
Content-Length: 206.
.
v=0.
o=CiscoSystemsSIP-GW-UserAgent 7438 1091 IN IP4 y.y.y.y.
s=SIP Call.
c=IN IP4 y.y.y.y.
t=0 0.
m=audio 18590 RTP/AVP 18.
c=IN IP4 y.y.y.y.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=yes.
#
U 2005/09/06 15:23:16.726095 a.b.c.d:5060 -> x.x.x.x:5060
ACK sip:x.x.x.x;ftag=c3ac7b24-a;lr=on SIP/2.0.
CSeq: 1 ACK.
Call-ID: call-00CC9E9D-4FBE-D311-0207-A at a.b.c.d.
Contact: <mailto:sip:046677 at a.b.c.d> <sip:046677 at a.b.c.d>.
From: <mailto:sip:046677 at a.b.c.d> <sip:046677 at a.b.c.d>;tag=c3ac7b24-a.
Route: <mailto:sip:0044123456789 at y.y.y.y:5060>
<sip:0044123456789 at y.y.y.y:5060>.
Session-GUID: 859321956-876164920-926430820-875782963.
To: <mailto:sip:123456789 at x.x.x.x>
<sip:123456789 at x.x.x.x>;tag=1A4297F4-2C9.
Via: SIP/2.0/UDP a.b.c.d;branch=z9hG4bK-tenor-c3ac-7b24-0018.
User-Agent: Quintum/1.0.0.
Quintum:
0c01030b0232360501000715000000000000000d006c0a828180803034363637371301000f0b
413031322d313033353941.
Max-Forwards: 70.
.
#
U 2005/09/06 15:23:16.727888 x.x.x.x:5060 -> y.y.y.y:5060
ACK sip:0044123456789 at y.y.y.y:5060 SIP/2.0.
Record-Route: <sip:x.x.x.x;ftag=c3ac7b24-a;lr=on>.
CSeq: 1 ACK.
Call-ID: call-00CC9E9D-4FBE-D311-0207-A at a.b.c.d.
Contact: <mailto:sip:046677 at a.b.c.d> <sip:046677 at a.b.c.d>.
From: <mailto:sip:046677 at a.b.c.d> <sip:046677 at a.b.c.d>;tag=c3ac7b24-a.
Session-GUID: 859321956-876164920-926430820-875782963.
To: <mailto:sip:123456789 at x.x.x.x>
<sip:123456789 at x.x.x.x>;tag=1A4297F4-2C9.
Via: SIP/2.0/UDP x.x.x.x;branch=0.
Via: SIP/2.0/UDP a.b.c.d;branch=z9hG4bK-tenor-c3ac-7b24-0018.
User-Agent: Quintum/1.0.0.
Quintum:
0c01030b0232360501000715000000000000000d006c0a828180803034363637371301000f0b
413031322d313033353941.
Max-Forwards: 16.
.
Jan Janak wrote:
Somehow the value from the first record-route got lost in the user
agent. It put the bottom most Route in the request-uri (which is correct
becuase the implemenetation is most likely a strict router), it also
correctly turned the Contact from 200 OK into Route header field in the
ACK, but the Route header field of the proxy (first one in 200 OK) is
missing in the ACK.
Jan.
On 08-09-2005 13:12, Iqbal wrote:
Hi Bogdan
Getting a little clearer :-)
The Contact that ius used is that from the 200 OK which is sent before
the ACK (as below), if so that explains where the .134 is coming from.
Also you mentioned that the last hop is taken, but I read somewhere that
the topmost route is taken, when forming the route.
If it is a strict router, does the RURI pick up the info from the
Contact header OR the RR,
Also should RURI not read 111.222.333.444 because that what is in the RR
in the 200 OK which is sent, which is what the ACK is showing. Whereas
in ur example it reads as below
b) if strict router:
RURI: <sip:212.156.5.135;ftag=c3ac7b24-a;lr=on>
R: <sip:111.222.333.444;ftag=c3ac7b24-a;lr=on>
R: <mailto:sip:0044123456789 at 212.156.5.134:5060>
<sip:0044123456789 at 212.156.5.134:5060>
Now if the RURI is as it is in the ACK, the rest should be
Route: <mailto:sip:0044123456789 at 212.156.5.135:5060>
<sip:0044123456789 at 212.156.5.135:5060>.
I guess at the end of the day whatever it should be, it isn't, anyone
know of howto fix it.
Iqbal
Bogdan-Andrei Iancu wrote:
Hi Iqbal,
the Route Set is formed by the all received RR uris and the Contact uri.
When doing LR, the last hop from the Route Set in loaded into RURI and
all RRs are loaded as Route. The request is not routed by RURI, but by
the first Route hdr.
So, if the GW does LR and it generates
RURI: <mailto:sip:0044123456789 at 212.156.5.134:5060>
<sip:0044123456789 at 212.156.5.134:5060>
R: <sip:212.156.5.135;ftag=c3ac7b24-a;lr=on>
R: <sip:111.222.333.444;ftag=c3ac7b24-a;lr=on>
,the request will be sent to the first Route (212.156.5.135), which
will sent to 111.222.333.444, which will finally use RURI
(212.156.5.134) as no other Route is left in request.
regards,
bogdan
Iqbal wrote:
Hi
This makes a little sense :-), according to a), is the path now taken
.444 --> .135, if so what does the RURI show.
If the path is that shown by Route headers, then I cannot send it to
.134, since that gateway does not respond directly, and will reject the
ACK.
Where does the RR, Route headers pick up the info from, I thought that
the Route gets it from what is in the RR
Iqbal
On 9/7/2005, "Bogdan-Andrei Iancu" <mailto:bogdan at voice-system.ro>
<bogdan at voice-system.ro> wrote:
Hi Iqbal,
considering 200 OK the Route Set:
RR: <sip:212.156.5.135;ftag=c3ac7b24-a;lr=on>
RR: <sip:111.222.333.444;ftag=c3ac7b24-a;lr=on>
CT: <mailto:sip:0044123456789 at 212.156.5.134:5060>
<sip:0044123456789 at 212.156.5.134:5060>
the ACK should be:
a)if loose router:
RURI: <mailto:sip:0044123456789 at 212.156.5.134:5060>
<sip:0044123456789 at 212.156.5.134:5060>
R: <sip:212.156.5.135;ftag=c3ac7b24-a;lr=on>
R: <sip:111.222.333.444;ftag=c3ac7b24-a;lr=on>
b) if strict router:
RURI: <sip:212.156.5.135;ftag=c3ac7b24-a;lr=on>
R: <sip:111.222.333.444;ftag=c3ac7b24-a;lr=on>
R: <mailto:sip:0044123456789 at 212.156.5.134:5060>
<sip:0044123456789 at 212.156.5.134:5060>
the ACK you posted looks like an unsuccessful attempt of a strict
routing.
regards,
bogdan
Iqbal wrote:
Hi
I have a strange looking dialogue, after sending a 200 OK to a quintum
box, and get back an ACK, but the Route field seems to be wrong, can
someone confirm this.
--------
U 2005/09/06 15:23:16.613003 111.222.333.444:5060 -> 10.20.30.40:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 10.20.30.40;branch=z9hG4bK-tenor-c3ac-7b24-0018.
From: <mailto:sip:046677 at 10.20.30.40>
<sip:046677 at 10.20.30.40>;tag=c3ac7b24-a.
To: <mailto:sip:123456789 at 111.222.333.444>
<sip:123456789 at 111.222.333.444>;tag=1A4297F4-2C9.
Date: Tue, 06 Sep 2005 14:07:59 gmt.
Call-ID: call-00CC9E9D-4FBE-D311-0207-A at 10.20.30.40.
Server: Cisco-SIPGateway/IOS-12.x.
CSeq: 1 INVITE.
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER,
SUBSCRIBE, NOTIFY, INFO.
Allow-Events: telephone-event.
Contact: <mailto:sip:0044123456789 at 212.156.5.134:5060>
<sip:0044123456789 at 212.156.5.134:5060>.
Record-Route:
<sip:212.156.5.135;ftag=c3ac7b24-a;lr=on>,<sip:111.222.333.444;ftag=c3ac7b24
-a;lr=on>.
Content-Type: application/sdp.
Content-Length: 206.
.
v=0.
o=CiscoSystemsSIP-GW-UserAgent 7438 1091 IN IP4 212.156.5.134.
s=SIP Call.
c=IN IP4 212.156.5.134.
t=0 0.
m=audio 18590 RTP/AVP 18.
c=IN IP4 212.156.5.134.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=yes.
#
U 2005/09/06 15:23:16.726095 10.20.30.40:5060 -> 111.222.333.444:5060
ACK sip:111.222.333.444;ftag=c3ac7b24-a;lr=on SIP/2.0.
CSeq: 1 ACK.
Call-ID: call-00CC9E9D-4FBE-D311-0207-A at 10.20.30.40.
Contact: <mailto:sip:046677 at 10.20.30.40> <sip:046677 at 10.20.30.40>.
From: <mailto:sip:046677 at 10.20.30.40>
<sip:046677 at 10.20.30.40>;tag=c3ac7b24-a.
Route: <mailto:sip:0044123456789 at 212.156.5.134:5060>
<sip:0044123456789 at 212.156.5.134:5060>.
Session-GUID: 859321956-876164920-926430820-875782963.
To: <mailto:sip:123456789 at 111.222.333.444>
<sip:123456789 at 111.222.333.444>;tag=1A4297F4-2C9.
Via: SIP/2.0/UDP 10.20.30.40;branch=z9hG4bK-tenor-c3ac-7b24-0018.
User-Agent: Quintum/1.0.0.
Quintum:
0c01030b0232360501000715000000000000000d006c0a828180803034363637371301000f0b
413031322d313033353941.
Max-Forwards: 70.
----------------
I thought the Route field should have the data which it pulls from the
Record Route headers
tks
iqbal
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