diff options
| author | Joshua Colp <jcolp@digium.com> | 2017-06-06 12:04:21 +0000 |
|---|---|---|
| committer | Joshua Colp <jcolp@digium.com> | 2017-06-07 13:34:58 +0000 |
| commit | d3e951edf5517b9f508a7e1b474176ec2be9e18f (patch) | |
| tree | a63f7b9110bad02f8cefca66ff8849a7d51c9ae9 /CHANGES | |
| parent | 9f054955f2f7830d4a7d20326d9fea7dff277456 (diff) | |
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
Diffstat (limited to 'CHANGES')
| -rw-r--r-- | CHANGES | 6 |
1 files changed, 6 insertions, 0 deletions
@@ -34,6 +34,12 @@ chan_pjsip function any contact which is considered unreachable due to qualify being enabled will no longer be called. + * The asymmetric_rtp_codec option now also controls whether chan_pjsip will + send media as-is without transcoding if the codec has been negotiated in the + SDP. If set to "no" then Asterisk will only ever send the preferred codec + from the SDP, unless the remote side sends a different codec and we will + switch to match. + ------------------------------------------------------------------------------ --- Functionality changes from Asterisk 14.4.0 to Asterisk 14.5.0 ------------ ------------------------------------------------------------------------------ |