[Asterisk-code-review] chan pjsip: Add support for multiple streams of the same type. (asterisk[master])

[Joshua Colp]

Hello Mark Michelson, Matthew Fredrickson, Richard Mudgett, Kevin Harwell, Jenkins2, 

I'd like you to reexamine a change. Please visit

    https://gerrit.asterisk.org/5876

to look at the new patch set (#16).

Change subject: chan_pjsip: Add support for multiple streams of the same type.
......................................................................

chan_pjsip: Add support for multiple streams of the same type.

The stream topology (list of streams and order) is now stored with the
configured PJSIP endpoints and used during the negotiation process.

Media negotiation state information has been changed to be stored
in a separate object. Two of these objects exist at any one time
on a session. The active media state information is what was previously
negotiated and the pending media state information is what the
media state will become if negotiation succeeds. Streams and other
state information is stored in this object using the index (or
position) of each individual stream for easy lookup.

The ability for a media type handler to specify a callback for
writing has been added as well as the ability to add file
descriptors with a callback which is invoked when data is available
to be read on them. This allows media logic to live outside of
the chan_pjsip module.

Direct media has been changed so that only the first audio and
video stream are directly connected. In the future once the RTP
engine glue API has been updated to know about streams each individual
stream can be directly connected as appropriate.

Media negotiation itself will currently answer all the provided streams
on an offer within configured limits and on an offer will use the
topology created as a result of the disallow/allow codec lines.

If a stream has been removed or declined we will now mark it as such
within the resulting SDP.

Applications can now also request that the stream topology change.
If we are told to do so we will limit any provided formats to the ones
configured on the endpoint and send a re-invite with the new topology.

Two new configuration options have also been added to PJSIP endpoints:

max_audio_streams: determines the maximum number of audio streams to
offer/accept from an endpoint. Defaults to 1.

max_video_streams: determines the maximum number of video streams to
offer/accept from an endpoint. Defaults to 1.

ASTERISK-27076

Change-Id: I8afd8dd2eb538806a39b887af0abd046266e14c7
---
M channels/chan_pjsip.c
M channels/pjsip/cli_commands.c
M channels/pjsip/dialplan_functions.c
M channels/pjsip/include/chan_pjsip.h
M configs/samples/pjsip.conf.sample
A contrib/ast-db-manage/config/versions/39959b9c2566_pjsip_stream_maximum.py
M include/asterisk/res_pjsip.h
M include/asterisk/res_pjsip_session.h
M include/asterisk/stream.h
M main/channel.c
M main/stream.c
M res/res_pjsip.c
M res/res_pjsip/pjsip_configuration.c
M res/res_pjsip_sdp_rtp.c
M res/res_pjsip_session.c
M res/res_pjsip_session.exports.in
M res/res_pjsip_t38.c
17 files changed, 1,895 insertions(+), 704 deletions(-)


  git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/76/5876/16
-- 
To view, visit https://gerrit.asterisk.org/5876
To unsubscribe, visit https://gerrit.asterisk.org/settings

Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-MessageType: newpatchset
Gerrit-Change-Id: I8afd8dd2eb538806a39b887af0abd046266e14c7
Gerrit-Change-Number: 5876
Gerrit-PatchSet: 16
Gerrit-Owner: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Jenkins2
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Kevin Harwell <kharwell at digium.com>
Gerrit-Reviewer: Mark Michelson <mmichelson at digium.com>
Gerrit-Reviewer: Matthew Fredrickson <creslin at digium.com>
Gerrit-Reviewer: Richard Mudgett <rmudgett at digium.com>
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