[Users] dialog module configuration question
Bogdan-Andrei Iancu
bogdan at voice-system.ro
Sun Feb 25 15:56:25 CET 2007
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 at 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 at 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 at 1.2.3.4:5060
> 9.) 1.2.3.4 -> 4.3.2.1 SIP Request: ACK sip:service at 4.3.2.1:5060
> 10.) 192.168.0.101 -> 1.2.3.4 SIP Request: BYE
> sip:service at 1.2.3.4:5060
> 11.) 1.2.3.4 -> 4.3.2.1 SIP Request: BYE sip:service at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 1.2.3.4:5060
>> > >> > > > 9.) 1.2.3.4 -> 4.3.2.1 SIP Request: ACK
>> > >> sip:service at 4.3.2.1:5060
>> > >> > > > 10.) 192.168.0.101 -> 1.2.3.4 SIP Request: BYE
>> > >> > > > sip:service at 1.2.3.4:5060
>> > >> > > > 11.) 1.2.3.4 -> 4.3.2.1 SIP Request: BYE
>> > >> sip:service at 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 at 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
>> > >> > > >> >
>> >
>>
>
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