[asterisk-dev] ast_sip_session

[Kelvin Chua]

thanks for the insights, i'm doing this as a proof of concept.
i relaized there are a lot of things which is a blocker by design.
what i initially tried is to get the bridged channel (of course this could
pose a problem when in multiples)
ast_channel_bridge_peer(session->channel)
then found out that this won't work because during this state, it's not yet
bridged.
looks like there is no way to get the ast_channel of the downstream while
call is not yet established, or is there?

on earlier versions, there is struct ast_channel *dialed and struct
ast_channel *dialing for this right?

On Wed, Nov 16, 2016 at 1:40 AM, Matthew Jordan <mjordan at digium.com> wrote:

>
>
> On Tue, Nov 15, 2016 at 6:17 AM, Joshua Colp <jcolp at digium.com> wrote:
>
>> On Mon, Nov 14, 2016, at 03:46 AM, Kelvin Chua wrote:
>> > playing around chan_pjsip/res_pjsip
>> >
>> > i have some data i want to pass from upstream to downstream on reponse
>> > for
>> > example, P-Associated-URI
>> >
>> > i figured that datastore on ast_sip_session only handles either the
>> > upstream channel or the downstream channel. I can already parse the
>> > header
>> > on upstream but stuck on how to write
>> > this to the reply downstream. any pointers?
>>
>> There is no generic mechanism for doing this currently: specifically
>> passing data back on a response or answer. The closest thing would be to
>> add a new frame type (frame.h) to Asterisk, put data on it (in an
>> ast_frame), propagate it through, and have it read/handled on the other
>> side. If you end up with specific questions you can respond back with
>> them, but you are sort of venturing into uncharted territory unless
>> someone else has an idea on how to better do it.
>>
>>
> I'd throw out there as well that this _may_ not be worth doing,
> particularly automatically. As a B2BUA, you're always going to be fighting
> against Asterisk's model of channels and bridges if you want to
> automatically propagate information from an inbound channel to an outbound
> channel.
>
> Things you have to think through:
> (1) What happens when there are Local channels?
> (2) What happens if there are multiple outbound channels (parallel dial)?
> (3) What happens if I'm in a bridge with multiple channels (conference)?
> (4) If the information can be provided in a re-INVITE, what occurs if I
> pass this information along before or after an answer?
>
> One of the safer ways of handling the notion of passing things from an
> inbound to an outbound channel is to have the dialplan do it. You can use
> functions like PJSIP_HEADER to read values off of an inbound channel, store
> them in a channel variable with inheritance, then use a pre-dial handler
> with PJSIP_HEADER (or CHANNEL, etc.) to modify the outbound channel(s)
> prior to dialing. Doing that gives you more control over exactly _when_ you
> pass information along, and helps you avoid the many, many edge cases that
> can occur.
>
> --
> Matthew Jordan
> Digium, Inc. | CTO
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at: http://digium.com & http://asterisk.org
>
> --
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