[asterisk-dev] [svn-commits] file: branch file/pjsip-subscription-persistence r415170 - /team/file/pjsip-s...

[Olle E. Johansson]

On 04 Jun 2014, at 20:57, Joshua Colp <jcolp at digium.com> wrote:

> Olle E. Johansson wrote:
>> On 04 Jun 2014, at 20:35, Joshua Colp<jcolp at digium.com>  wrote:
>> 
>>> Olle E. Johansson wrote:
>>>> On 04 Jun 2014, at 19:49, SVN commits to the Digium
>>>> repositories<svn-commits at lists.digium.com>   wrote:
>>>> 
>>>>> Creating a branch for persisting subscriptions across restarts
>>>>> in PJSIP.
>>>> Why? You can terminate subscriptions at shutdown and the UA will
>>>> re-subscribe when you're back.
>>> The goal of my branch is to cover both best cases and worst cases.
>>> In normal circumstances then yes, we can terminate with probation
>>> or deactivated and assume the other side will re-subscribe. In
>>> worst case Asterisk has terminated unexpectedly (or "core restart
>>> now" is used) and we don't have a chance to do this. I would like
>>> the experience for the user to be that Asterisk comes back and they
>>> are subscribed, persistence achieves this in both the best and
>>> worst case scenarios.
>> Well, yes. Agree.
>> 
>> Remember the hook we have so that we terminate subscriptions for
>> extensions that was deleted when the dialplan was reloaded.
> 
> Will do.
> 
>>>> The problem with registrations is that you can not do that.
>>> Can you clarify what you mean by this?
>> 
>> You can not terminate a registration from the server. If you do
>> support SUBSCRIBE for registration states you can do a bit more
>> messaging to the UA, which would be a good thing to support,
>> especially since you want to support multiple registrations per
>> account.
> 
> Do you know off the top of your head any endpoints which are easily available which implement this for testing?

No, not on the top of my head. I will check. Have been discussing this a lot with Blink.

It makes a lot of sense to do it. Phones will know which other phones
that are connected to an account and can simplify transfer, especially
if their contacts contact display names.

A digium phone supporting this could have a quick transfer menu
saying something like

"transfer to
Olles laptop
Olles supervideoscreen
Olles audio-only BT phone"

Then I could easily transfer a call from one SIP phone to another using my account. If they all use GRUUs,
IPv4/IPv6/NAT won't be a big issue either.

Sorry for preaching ;-)


/O