diff --git a/CHANGES b/CHANGES index a951b2ca393b24ee589180667379f60c3e096ed9..5cfffda155f8a40a0adb17b10d7ef94b3402144f 100644 --- a/CHANGES +++ b/CHANGES @@ -34,6 +34,12 @@ 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 6eea7936815eda9ec3d21377cc90a123d2b4e763..486a237aed5939737ef26db21800d378a8b266a9 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 cafbd52ec83e2e4fa381bf290343c9e9a5ba18a9..d39842f3a8fb726b79beb7aeb13873ff327baa7d 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