[Asterisk-code-review] ACN: Advanced Codec Negotiation for chan_pjsip (asterisk[master])

[George Joseph]

Hello Friendly Automation, 

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

    https://gerrit.asterisk.org/c/asterisk/+/14637

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

Change subject: ACN: Advanced Codec Negotiation for chan_pjsip
......................................................................

ACN: Advanced Codec Negotiation for chan_pjsip

This commit is the second in a series that implements
Advanced Codec Negotiation and does not represent a final
implementation.  Many existing features either do not yet work
or have yet to be tested including...

 * Direct media
 * 100rel/early media
 * Re-invites
 * Fax
 * Multi-stream
 * Deferred SDP
 * ARI channel operations
 * Operation with other channel technologies

There are also several refactors that still need to happen to remove
duplicated code, re-organize internal functions, consolidate
activities, etc.

Summary of functional changes by module:

res_pjsip
 * Updated documentation.
 * Removed obsolete code.

res_pjsip/pjsip_configuration
 * Updated endpoint configuration parsing.
 * Removed obsolete code.

res_pjsip_refer
 * Use ast_raw_answer_with_stream_topology instead of ast_raw_answer.

res_pjsip_sdp_rtp
 * Refactored get_codecs to return an ast_format_caps.
 * Removed obsolete code in set_incoming_call_offer_cap.
 * Removed obsolete code in set_caps.
 * Removed obsolete code in negotiate_incoming_sdp_stream.
 * Removed obsolete code in create_outgoing_sdp_stream.

res_pjsip_session
 * Preserve the pending topology in handle_negotiated_sdp so
   chan_pjsip can do a topology resolution.
 * Removed obsolete code in ast_sip_session_create_outgoing.
 * Removed obsolete code in new_invite.
 * Updated session_inv_on_tsx_state_changed to call handlers with
   the proper BEFORE/AFTER MEDIA flag.
 * Removed obsolete code in create_local_sdp.

res_pjsip_session/pjsip_session_caps
tests/test_res_pjsip_session_caps
 * Removed.

chan_pjsip
 * Implement chan_pjsip_answer_with_stream_topology
 * chan_pjsip_incoming_request now resolves the topology on an
   incoming invite with that of the endpoint.
 * chan_pjsip_request_with_stream_topology now resolves the
   topology passed from the core on an outgoing call with that
   of the endpoint.
 * chan_pjsip_incoming_response now resolves the topology received
   on a 200 OK with that which was sent on the original invite.
 * chan_pjsip_answer_with_stream_topology now resolves the topology
   passed from the core with that which was originally passed to
   the core.

ASTERISK-28856

Change-Id: Iad188ae997bdcb5c28e2eb12c6bb2b732538ad45
---
M channels/chan_pjsip.c
M configs/samples/pjsip.conf.sample
M contrib/ast-db-manage/config/versions/b80485ff4dd0_add_pjsip_endpoint_acn_options.py
M include/asterisk/res_pjsip.h
M include/asterisk/res_pjsip_session.h
D include/asterisk/res_pjsip_session_caps.h
M main/bridge_channel.c
M res/Makefile
M res/res_pjsip.c
M res/res_pjsip/pjsip_configuration.c
M res/res_pjsip_refer.c
M res/res_pjsip_sdp_rtp.c
M res/res_pjsip_session.c
D res/res_pjsip_session/pjsip_session_caps.c
D tests/test_res_pjsip_session_caps.c
15 files changed, 768 insertions(+), 1,194 deletions(-)


  git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/37/14637/2
-- 
To view, visit https://gerrit.asterisk.org/c/asterisk/+/14637
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Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-Change-Id: Iad188ae997bdcb5c28e2eb12c6bb2b732538ad45
Gerrit-Change-Number: 14637
Gerrit-PatchSet: 2
Gerrit-Owner: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: Friendly Automation
Gerrit-CC: Joshua Colp <jcolp at sangoma.com>
Gerrit-MessageType: newpatchset
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