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
This commit is contained in:
Joshua Colp
2017-06-06 12:04:21 +00:00
parent 812f5b51cb
commit 996a4791ff
3 changed files with 38 additions and 2 deletions

View File

@@ -401,7 +401,24 @@ static int set_caps(struct ast_sip_session *session, struct ast_sip_session_medi
ast_format_cap_append_from_cap(caps, ast_channel_nativeformats(session->channel),
AST_MEDIA_TYPE_UNKNOWN);
ast_format_cap_remove_by_type(caps, media_type);
ast_format_cap_append_from_cap(caps, joint, media_type);
/*
* If we don't allow the sending codec to be changed on our side
* then get the best codec from the joint capabilities of the media
* type and use only that. This ensures the core won't start sending
* out a format that we aren't currently sending.
*/
if (!session->endpoint->asymmetric_rtp_codec) {
struct ast_format *best;
best = ast_format_cap_get_best_by_type(joint, media_type);
if (best) {
ast_format_cap_append(caps, best, ast_format_cap_get_framing(joint));
ao2_ref(best, -1);
}
} else {
ast_format_cap_append_from_cap(caps, joint, media_type);
}
/*
* Apply the new formats to the channel, potentially changing