[Asterisk-code-review] chan pjsip: Multistream: Use underlying multistream structures. (asterisk[master])

[Mark Michelson]

Hello Jenkins2,

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

    https://gerrit.asterisk.org/5760

to look at the new patch set (#2).

Change subject: chan_pjsip: Multistream: Use underlying multistream structures.
......................................................................

chan_pjsip: Multistream: Use underlying multistream structures.

This commit is step one towards making chan_pjsip capable of supporting
multiple streams of a given type. The basic thing that's been done here
is to make it so that rather than using ast_format_cap structures in
places, we instead are using ast_stream_topology and ast_stream. There
is no functionality change being made by this commit; you still can only
have a single audio and single video stream on a call with a PJSIP
endpoint.

Throughout this commit some XXX comments have been added in places where
some assumptions may no longer hold when actually making the code
support multistream.

Change-Id: I0358ea3108e2c06ae43308abddf216a23deccb90
---
M channels/chan_pjsip.c
M channels/pjsip/dialplan_functions.c
M include/asterisk/res_pjsip.h
M include/asterisk/res_pjsip_session.h
M include/asterisk/stream.h
M res/res_pjsip/pjsip_configuration.c
M res/res_pjsip_sdp_rtp.c
M res/res_pjsip_session.c
8 files changed, 243 insertions(+), 78 deletions(-)


  git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/60/5760/2
-- 
To view, visit https://gerrit.asterisk.org/5760
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Gerrit-MessageType: newpatchset
Gerrit-Change-Id: I0358ea3108e2c06ae43308abddf216a23deccb90
Gerrit-PatchSet: 2
Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-Owner: Mark Michelson <mmichelson at digium.com>
Gerrit-Reviewer: Jenkins2