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-rw-r--r--CHANGES6
-rw-r--r--channels/chan_pjsip.c15
-rw-r--r--res/res_pjsip_sdp_rtp.c16
3 files changed, 36 insertions, 1 deletions
diff --git a/CHANGES b/CHANGES
index 829086513..9bfa506d3 100644
--- a/CHANGES
+++ b/CHANGES
@@ -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 ------------
------------------------------------------------------------------------------
diff --git a/channels/chan_pjsip.c b/channels/chan_pjsip.c
index 3f65a13de..83dc77f38 100644
--- a/channels/chan_pjsip.c
+++ b/channels/chan_pjsip.c
@@ -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);
diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c
index 97e365c10..c5a673aa4 100644
--- a/res/res_pjsip_sdp_rtp.c
+++ b/res/res_pjsip_sdp_rtp.c
@@ -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.