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 9f054955f2
commit d3e951edf5
3 changed files with 36 additions and 1 deletions

View File

@@ -34,6 +34,12 @@ chan_pjsip
function any contact which is considered unreachable due to qualify being function any contact which is considered unreachable due to qualify being
enabled will no longer be called. 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 ------------ --- Functionality changes from Asterisk 14.4.0 to Asterisk 14.5.0 ------------
------------------------------------------------------------------------------ ------------------------------------------------------------------------------

View File

@@ -735,11 +735,24 @@ static struct ast_frame *chan_pjsip_read(struct ast_channel *ast)
if (!session->endpoint->asymmetric_rtp_codec && if (!session->endpoint->asymmetric_rtp_codec &&
ast_format_cmp(ast_channel_rawwriteformat(ast), f->subclass.format) == AST_FORMAT_CMP_NOT_EQUAL) { ast_format_cmp(ast_channel_rawwriteformat(ast), f->subclass.format) == AST_FORMAT_CMP_NOT_EQUAL) {
/* For maximum compatibility we ensure that the write format matches that of the received media */ struct ast_format_cap *caps;
/* For maximum compatibility we ensure that the formats match that of the received media */
ast_debug(1, "Oooh, got a frame with format of %s on channel '%s' when we're sending '%s', switching to match\n", ast_debug(1, "Oooh, got a frame with format of %s on channel '%s' when we're sending '%s', switching to match\n",
ast_format_get_name(f->subclass.format), ast_channel_name(ast), ast_format_get_name(f->subclass.format), ast_channel_name(ast),
ast_format_get_name(ast_channel_rawwriteformat(ast))); ast_format_get_name(ast_channel_rawwriteformat(ast)));
caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
if (caps) {
ast_format_cap_append_from_cap(caps, ast_channel_nativeformats(ast), AST_MEDIA_TYPE_UNKNOWN);
ast_format_cap_remove_by_type(caps, AST_MEDIA_TYPE_AUDIO);
ast_format_cap_append(caps, f->subclass.format, 0);
ast_channel_nativeformats_set(ast, caps);
ao2_ref(caps, -1);
}
ast_set_write_format_path(ast, ast_channel_writeformat(ast), f->subclass.format); ast_set_write_format_path(ast, ast_channel_writeformat(ast), f->subclass.format);
ast_set_read_format_path(ast, ast_channel_readformat(ast), f->subclass.format);
if (ast_channel_is_bridged(ast)) { if (ast_channel_is_bridged(ast)) {
ast_channel_set_unbridged_nolock(ast, 1); ast_channel_set_unbridged_nolock(ast, 1);

View File

@@ -410,13 +410,29 @@ static int set_caps(struct ast_sip_session *session,
ast_format_cap_append_from_cap(caps, ast_channel_nativeformats(session->channel), ast_format_cap_append_from_cap(caps, ast_channel_nativeformats(session->channel),
AST_MEDIA_TYPE_UNKNOWN); AST_MEDIA_TYPE_UNKNOWN);
ast_format_cap_remove_by_type(caps, media_type); ast_format_cap_remove_by_type(caps, media_type);
if (session->endpoint->preferred_codec_only){ if (session->endpoint->preferred_codec_only){
struct ast_format *preferred_fmt = ast_format_cap_get_format(joint, 0); struct ast_format *preferred_fmt = ast_format_cap_get_format(joint, 0);
ast_format_cap_append(caps, preferred_fmt, 0); ast_format_cap_append(caps, preferred_fmt, 0);
ao2_ref(preferred_fmt, -1); ao2_ref(preferred_fmt, -1);
} else if (!session->endpoint->asymmetric_rtp_codec) {
struct ast_format *best;
/*
* 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.
*/
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 { } else {
ast_format_cap_append_from_cap(caps, joint, media_type); ast_format_cap_append_from_cap(caps, joint, media_type);
} }
/* /*
* Apply the new formats to the channel, potentially changing * Apply the new formats to the channel, potentially changing
* raw read/write formats and translation path while doing so. * raw read/write formats and translation path while doing so.