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

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Jenkins2
2017-06-13 09:17:10 -05:00
committed by Gerrit Code Review
parent 5d3420a2de d3e951edf5
commit abe1dd3039
3 changed files with 36 additions and 1 deletions
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6
CHANGES
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@@ -43,6 +43,12 @@ 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 ------------
------------------------------------------------------------------------------

15
channels/chan_pjsip.c
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@@ -735,11 +735,24 @@ static struct ast_frame *chan_pjsip_read(struct ast_channel *ast)
if (!session->endpoint->asymmetric_rtp_codec &&
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_format_get_name(f->subclass.format), ast_channel_name(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_read_format_path(ast, ast_channel_readformat(ast), f->subclass.format);
if (ast_channel_is_bridged(ast)) {
ast_channel_set_unbridged_nolock(ast, 1);

16
res/res_pjsip_sdp_rtp.c
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@@ -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_MEDIA_TYPE_UNKNOWN);
ast_format_cap_remove_by_type(caps, media_type);
if (session->endpoint->preferred_codec_only){
struct ast_format *preferred_fmt = ast_format_cap_get_format(joint, 0);
ast_format_cap_append(caps, preferred_fmt, 0);
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 {
ast_format_cap_append_from_cap(caps, joint, media_type);
}
/*
* Apply the new formats to the channel, potentially changing
* raw read/write formats and translation path while doing so.