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-rw-r--r--CHANGES6
-rw-r--r--channels/chan_pjsip.c15
-rw-r--r--res/res_pjsip_sdp_rtp.c19
3 files changed, 38 insertions, 2 deletions
diff --git a/CHANGES b/CHANGES
index 2dd47245d..6007928df 100644
--- a/CHANGES
+++ b/CHANGES
@@ -25,6 +25,12 @@ 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 13.15.0 to Asterisk 13.16.0 ----------
------------------------------------------------------------------------------
diff --git a/channels/chan_pjsip.c b/channels/chan_pjsip.c
index 97c3d101b..fdbeef8aa 100644
--- a/channels/chan_pjsip.c
+++ b/channels/chan_pjsip.c
@@ -737,11 +737,24 @@ 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 cafbd52ec..d39842f3a 100644
--- a/res/res_pjsip_sdp_rtp.c
+++ b/res/res_pjsip_sdp_rtp.c
@@ -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