25 Feb
2007
25 Feb
'07
5:56 p.m.
Hi Andy,
yes, you are correct - the package 7 (the 200 OK) must mirror the
Record-Route set from the request. For this you need to enable the rrs
param :
<recv request="INVITE" crlf="true" rrs="true">
</recv>
regards,
bogdan
Andy Pyles wrote:
Hi Bogdan,
ok let me go back to my example:
Here's more detail:
192.168.0.101 = Caller (sipp uas)
1.2.3.4 = openser
4.3.2.1 = callee ( sipp uac)
1.) 192.168.0.101 -> 1.2.3.4 SIP/SDP Request: INVITE
sip:service@1.2.3.4:5060, with session description
2.) 1.2.3.4 -> 192.168.0.101 SIP Status: 100 Giving a try
3.) 1.2.3.4 -> 4.3.2.1 SIP/SDP Request: INVITE
sip:service@4.3.2.1:5060, with session description
4.) 4.3.2.1 -> 1.2.3.4 SIP Status: 180 Ringing
5.) 4.3.2.1 -> 1.2.3.4 SIP/SDP Status: 200 OK, with session
description
6.) 1.2.3.4 -> 192.168.0.101 SIP Status: 180 Ringing
7.) 1.2.3.4 -> 192.168.0.101 SIP/SDP Status: 200 OK, with session
description
8.) 192.168.0.101 -> 1.2.3.4 SIP Request: ACK
sip:service@1.2.3.4:5060
9.) 1.2.3.4 -> 4.3.2.1 SIP Request: ACK sip:service@4.3.2.1:5060
10.) 192.168.0.101 -> 1.2.3.4 SIP Request: BYE
sip:service@1.2.3.4:5060
11.) 1.2.3.4 -> 4.3.2.1 SIP Request: BYE sip:service@4.3.2.1:5060
12.) 4.3.2.1 -> 1.2.3.4 SIP Status: 200 OK
13.) 1.2.3.4 -> 192.168.0.101 SIP Status: 200 OK
So, you are saying for Packets 8, 10 I should add the '[routes]' logic
to sipp. How this works is: from the sipp documentation: "rrs: Record
Route Set. if this attribute is set to "true", then the
"Record-Route:" header of the message received is stored and can be
recalled using the [routes] keyword.".
This I completey agree with. sipp Must be sending the Route: header
in Packets 8 and 10. However, packet 7 MUST have the Record-route
header, otherwise, How can sipp can put the correct value into the
Route: header. See my point?
Reference: rfc 3665 ( secion 3.2 Packet f11, f14)
regards,
Andy
On 2/23/07, Andy Pyles <andy.pyles(a)gmail.com> wrote:
Hi Bogdan,
correct. but on client config "[routes]" ( for sipp) will only work
IF the client receives a Record-route. Since I'm not, it doesn't help
me. Am I missing something?
Andy
On 2/23/07, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> wrote:
Hi Andy,
in client config, you need to add "[routes]" for ACK and BYE messages
(take a look at the cfg I sent you)
regards,
bogdan
Andy Pyles wrote:
> I Just re-read the docs on loose_route(). So please disregard this
> question. ( only processed if Route: header is present. Which isn't
> present because Record-route: header isn't being sent to caller )
>
> So, I'm still trying to figure out why record-route: header is not
> being sent to caller.
>
>
> On 2/22/07, Andy Pyles <andy.pyles(a)gmail.com> wrote:
>> Hi Bogdan,
>>
>> After running additional debugs, for some reason the call to
>> loose_route() is failing.
>>
>> if (loose_route()) {
>> # mark routing logic in request
>> xlog("L_INFO", "loose_route() succeeded\n ");
>> route(1);
>> } else{
>> xlog("L_INFO", "loose_route()failed - M=$rm RURI=$ru
F=$fu
>> T=$tu IP=$si ID=$ci\n");
>> };
>>
>>
>> Any ideas why this could be occuring?
>>
>>
>> On 2/22/07, Andy Pyles <andy.pyles(a)gmail.com> wrote:
>> > HI Bogdan,
>> >
>> > I'm already using an almsot identical version of uas.xml and
uac.xml (
>> > yes rrs=true) is being used.
However in your version the uas.xml
>> > doesn't have rrs="true" after initial invite which I think is
needed.
>> > See as you can see below, setting
rrs="true" for uac will only
work if
>> > it receives a Record-Route header
in the 200OK which it's not.
>> >
>> > In this case, ALL messages from openser to sipp uac do not
contain
the
>> > Record-route header. So I don't
think it's a sipp problem, but an
>> > openser configuration problem. I've tried using other devices
for a
>> > uac, such as x-lite but the same
problem.
>> >
>> > Andy
>> >
>> > On 2/22/07, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> wrote:
>> > > Hi Andy,
>> > >
>> > > so it's about sipp :D - I remember I had some hard times to
make
>> it work
>> > > with record Route.
>> > >
>> > > take a look at the attached files, they might help you.
>> > >
>> > > regards,
>> > > bogdan
>> > >
>> > > Andy Pyles wrote:
>> > > > HI Bogdan,
>> > > >
>> > > > thanks for your reply.
>> > > > yes you are correct. The Bye doesn't have the Route header.
>> > > > It appears the the 200 OK sent to the caller doesn't
contain a
>> > > > Record-route header.
>> > > > Messages between openser and callee contain record-route
>> information,
>> > > > but messages between caller and openser do not.
>> > > > Is there a way to enable that?
>> > > >
>> > > > Here's more detail:
>> > > > 192.168.0.101 = Caller (sipp)
>> > > > 1.2.3.4 = openser
>> > > > 4.3.2.1 = callee ( sipp)
>> > > >
>> > > >
>> > > > 1.) 192.168.0.101 -> 1.2.3.4 SIP/SDP Request: INVITE
>> > > > sip:service@1.2.3.4:5060, with session description
>> > > > 2.) 1.2.3.4 -> 192.168.0.101 SIP Status: 100 Giving a try
>> > > > 3.) 1.2.3.4 -> 4.3.2.1 SIP/SDP Request: INVITE
>> > > > sip:service@4.3.2.1:5060, with session description
>> > > > 4.) 4.3.2.1 -> 1.2.3.4 SIP Status: 180 Ringing
>> > > > 5.) 4.3.2.1 -> 1.2.3.4 SIP/SDP Status: 200 OK, with
>> session
>> > > > description
>> > > > 6.) 1.2.3.4 -> 192.168.0.101 SIP Status: 180 Ringing
>> > > > 7.) 1.2.3.4 -> 192.168.0.101 SIP/SDP Status: 200 OK, with
>> session
>> > > > description
>> > > > 8.) 192.168.0.101 -> 1.2.3.4 SIP Request: ACK
>> > > > sip:service@1.2.3.4:5060
>> > > > 9.) 1.2.3.4 -> 4.3.2.1 SIP Request: ACK
>> sip:service@4.3.2.1:5060
>> > > > 10.) 192.168.0.101 -> 1.2.3.4 SIP Request: BYE
>> > > > sip:service@1.2.3.4:5060
>> > > > 11.) 1.2.3.4 -> 4.3.2.1 SIP Request: BYE
>> sip:service@4.3.2.1:5060
>> > > > 12.) 4.3.2.1 -> 1.2.3.4 SIP Status: 200 OK
>> > > > 13.) 1.2.3.4 -> 192.168.0.101 SIP Status: 200 OK
>> > > >
>> > > > ---
>> > > > Packets 6,7 and following contain no Record-route
information.
>> > > > The other weird thing is
that openser is passing on the
Route:
>> header
>> > > > it recevied from callee to the caller.
>> > > >
>> > > >
>> > > > Please see attached for complete ngrep output.
>> > > >
>> > > >
>> > > > On 2/21/07, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro>
wrote:
>> > > >> Hi Andy,
>> > > >>
>> > > >> could you check on the net if the BYE contain the Route hdr
>> added to
>> > > >> INVITE as Record-Route? I have some doubts on this as I see:
>> > > >> 0(966) find_first_route: No Route headers found
>> > > >> 0(966) loose_route: There is no Route HF
>> > > >>
>> > > >> and if the BYE is not identified, the dialog is not closed.
>> > > >>
>> > > >> regards,
>> > > >> bogdan
>> > > >>
>> > > >> Andy Pyles wrote:
>> > > >> > Hello,
>> > > >> >
>> > > >> > I have a question on how to configure the dialog module
(
>> 1.2.x from
>> > > >> > cvs yesterday ).
>> > > >> >
>> > > >> > With my config, ( attached) I can make calls and have
>> verified that
>> > > >> > the acc module is working correctly.
>> > > >> >
>> > > >> > My question is, when I enable the dialog module, I can
see
>> that it is
>> > > >> > incrementing call count correctly, but when a bye is
>> received, the
>> > > >> > dialog:active_dialogs statistic is never decremented.
>> > > >> >
>> > > >> > In the debug level 9 logs, ( also attached) I see this
error
>> after the
>> > > >> > 200OK is sent to the bye:
>> > > >> >
>> > > >> > 1(969) DBUG:dialog:unref_dlg: unref dlg 0xa7ce5a98 with
1
>> > > >> (delete=0)-> 1
>> > > >> >
>> > > >> > Is this a case of one of the timers being set too
short? by
> the
way
> > > >> > using a variable call length from well under a second (
> using sipp )
> > > >> > to 20 second call doesnt' seem to make a difference .
> > > >> >
> > > >> >
> > > >> > Thanks,
> > > >> > Andy
> > > >> >