[SR-Users] Wrong ACK to Provider

[Daniel-Constantin Mierla]

On 28/08/14 14:45, Olle E. Johansson wrote:
>
> On 28 Aug 2014, at 14:14, Yuriy Gorlichenko <ovoshlook at gmail.com 
> <mailto:ovoshlook at gmail.com>> wrote:
>
>> Hello. I try to provide call scheme:
>>
>> internal client  -> asterisk -> Kamailio -> provider -> external 
>> endpoint call
>>
>> when I make call I see this:
>>
>> asterisk     kamailio   provider
>> invite -->       invite -->
>>                                 <--     407
>>                        ACK   -->
>>                        invite w/Auth -->
>>               <--    100  <--    100
>>               <--    180  <--    180
>>               <--    183  <--    183
>>                <--    200  <--      200
>>    ACK  -->   ACK  -->
>>
>> My problem with last ACK, that I send to provider. Provider ignores 
>> it, and sends me some OK packets. As resultI can notend session ( 
>> answer to BYE 481 - transaction does not exists). I think it is wrong 
>> ACK but can not undrtand where I do mistake.
> Well, by letting the proxy handle authentication the INVITE tranction 
> i closed without Asterisk knowing about it. So the ACK sent from the 
> proxy and from Asterisk is for the same transaction, which messes 
> things up. Asterisk does not know anything about the second invite. 
> Letting the proxy handle authentiction breaks the SIP protocol in bad 
> ways and is generally not a good solution.
> You may want to send another response to asterisk when you get the 407 
> so Asterisk retries and use the retry as a trigger for the second 
> INVITE and add auth to that.
While breaking the cseq incrementation for authentication (mentioned in 
the readme of uac), the Asterisk seems to do ok here, because the ACK is 
coming from asterisk, but it is not accepted by the provider.

The provider (having a plivo platform, based on the responses) is 
running kamailio 4.1.2 in front (looking at 100 trying).

Authentication from kamailio to another kamailio using uac module should 
work fine, as kamailio doesn't act as end user UAC and doesn't care much 
of cseq.

I didn't have time to look at the sip trace properly, but Asterisk 
should have nothing to do with the problem here, unless I missed 
something from the description.

Cheers,
Daniel


-- 
Daniel-Constantin Mierla
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