Ivar wrote:
>
>
> >If it didn't do that, and it subsequently originates a BYE then it is
> simply violating the specs.
> Hmm, but can b2bua not alter SDP and not play as RTP proxy, only SIP.
> (I mean not carrying audio through SIP proxy)
Yes, it can be quite reasonable for a B2BUA to only participate in the
sip signaling, not the media. If its careful how it does things it need
not touch the SDP at all. In other cases it may want to manipulate the
SDP, but in such a way that it still doesn't ever terminate the media.
> Are there some good b2bua docs what describes it, RFC 3261 isn't very
> helpfull.
You could look at draft-marjou-sipping-b2bua-00.
For the most part this subject is sort of the "wild west", with people
doing pretty much whatever they want.
Paul
> Thanks,
>
> Paul Kyzivat wrote:
>>
>>
>> Ivar wrote:
>>> Hi,
>>>
>>> Can somebody suggest me some docs related to call statefull proxy
>>> (not b2bua) ?
>>
>> AFAIK there is not much written about this. Quite frankly, call
>> stateful proxies just aren't very useful.
>>
>>> If i get right normally call-statefull proxy is just used to get call
>>> duration and possibility terminate call by sending BYE to caller and
>>> callee ?
>>
>> A call stateful *proxy* is not permitted to originate a BYE for the
>> session. That is a UA function. If that is done, then the server is
>> acting as a B2BUA. To be a UA it would have to use its own contact
>> address during session establishment. If it didn't do that, and it
>> subsequently originates a BYE then it is simply violating the specs.
>>
>>> Or i miss something. Anyway thats functionality what i would like to
>>> implement.
>>>
>>> 1) Call state is done with 2xx response to INVITE.
>>> 2) Call is ended by BYE request from caller or calee.
>>> 3) No BYE, probably session timer will do the call termination ? I
>>> haven't looked it yet.
>>
>> You can do this. If so, the proxy must Record-Route so that it stays
>> in the signaling path for the duration of the session. It is then
>> dependent on one or the other of the UAs in the call supporting
>> session timer. If neither does, then the proxy has a problem.
>>
>>> Call
>>> authUserName
>>> to:
>>> from:
>>> startTime:
>>> Terminate()
>>>
>>> Is each RTP session as new call or call must keep track RTP sessions
>>> too.
>>> (If must, probably only b2bua can do this, not call statefull)
>>
>> I don't understand your question. During the call the media being used
>> can be renegotiated using reINVITE or UPDATE. This is still a single
>> sip session, but if you are doing billing, you might want to change it.
>>
>> In general it is problematic to generate billing solely from the
>> signaling.
>>
>> Paul
>
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