On 06/05/2017 03:17 PM, Matt Fredrickson wrote:
On Mon, Jun 5, 2017 at 2:31 PM, Joshua Colp <jc...@digium.com> wrote:
On Mon, Jun 5, 2017, at 04:21 PM, Mark Michelson wrote:
Hi folks,

For those of you following along at home, I recently published review
https://gerrit.asterisk.org/#/c/5760/ , which is step one towards making
chan_pjsip multistream, i.e. supporting more than one stream of a given
media type. This initial review does not actually introduce multistream
support so much as it just makes use of multistream structures under the
hood to ease the transition.

Continuing on, one of the next steps we need to determine is how a user
of chan_pjsip should configure a channel that supports multiple streams
of a particular type.

To refresh everyone on how things currently work in pjsip.conf, you set
an "allow" option in order to determine what codecs a particular
endpoint supports.

[Alice]
type = endpoint
allow = ulaw,opus,h264

<snip>

I propose the following configuration options to move forward.

offered_audio_streams = 1
offered_video_streams = 2
<snip>

My only question is why, in a scenario where we don't have a hint, would
we want to make the number of offered streams configurable by the user?
Ultimately it's up to the application that is handling the channel to
decide what it wants and that is decided in the moment, not ahead of
time based on configuration.

I think maximum and minimum are useful for enforcing some constraints
though.
That echoes my thoughts as well.  +1 to not having the
"offered_audio/video_streams" options, but I'm ok with the limiting of
maximum stream count for now.

Cool. I'm all for having fewer options. Sounds like if a user wants a certain number of streams to be offered during origination, that should be a parameter for the origination, then.


--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-dev

Reply via email to