[Asterisk-code-review] pjsip: Extend 'asymmetric rtp codec' option to include us ch... (asterisk[13])

[Joshua Colp]

Hello George Joseph, Jenkins2,

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

    https://gerrit.asterisk.org/5765

to look at the new patch set (#3).

Change subject: pjsip: Extend 'asymmetric_rtp_codec' option to include us changing.
......................................................................

pjsip: Extend 'asymmetric_rtp_codec' option to include us changing.

PJSIP support in Asterisk differs from chan_sip in that it
allows media to be sent as-is without transcoding provided
the codecs were negotiated in the SDP. This is allowed
according to the RFC. Support for this differs quite a lot
though and some endpoints do not handle it well.

This change extends the 'asymmetric_rtp_codec' option to
also cover this case. When set to no (the default) the code
behaves as chan_sip does - the best codec is selected and
we will only ever send that, unless we change what we are
sending if the remote side changes. When set to yes we
will send media as-is without transcoding if the codec
has been negotiated in the SDP.

ASTERISK-26996

Change-Id: Ib1647f6902a0843e8c435946f831c2159e8d1d51
---
M CHANGES
M channels/chan_pjsip.c
M res/res_pjsip_sdp_rtp.c
3 files changed, 38 insertions(+), 2 deletions(-)


  git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/65/5765/3
-- 
To view, visit https://gerrit.asterisk.org/5765
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Gerrit-MessageType: newpatchset
Gerrit-Change-Id: Ib1647f6902a0843e8c435946f831c2159e8d1d51
Gerrit-PatchSet: 3
Gerrit-Project: asterisk
Gerrit-Branch: 13
Gerrit-Owner: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: Jenkins2