diff options
| author | Matthew Jordan <mjordan@digium.com> | 2015-03-02 19:15:58 +0000 |
|---|---|---|
| committer | Matthew Jordan <mjordan@digium.com> | 2015-03-02 19:15:58 +0000 |
| commit | 278ea2f468145a6a187743882c39241a999691e5 (patch) | |
| tree | 13ab824ac54df9d37e931010b9303f44c13c1389 | |
| parent | 9e841e4fb62f072f26f076bd4dd2a8eebe7a9e75 (diff) | |
res/res_pjsip_sdp_rtp: Revert portion of r432195
Unfortunately, while initial testing with ConfBridge did not reproduce the
audio problem alluded to in the comment in res_pjsip_sdp_rtp, further testing
did show that bridge_softmix and/or ConfBridge has a severe problem bridging
two or more participants at different sampling rates. Sometimes, it even picks
odd sampling rates that cause hideous audio problems.
This patch backs out the offending portion of the code until the issues in
the affected bridging modules can be more properly analyzed.
ASTERISK-24841
........
Merged revisions 432423 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432425 65c4cc65-6c06-0410-ace0-fbb531ad65f3
| -rw-r--r-- | res/res_pjsip_sdp_rtp.c | 25 |
1 files changed, 22 insertions, 3 deletions
diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c index 3ec61e1e5..a2550df9e 100644 --- a/res/res_pjsip_sdp_rtp.c +++ b/res/res_pjsip_sdp_rtp.c @@ -240,8 +240,8 @@ static int set_caps(struct ast_sip_session *session, struct ast_sip_session_medi /* get the joint capabilities between peer and endpoint */ ast_format_cap_get_compatible(caps, peer, joint); if (!ast_format_cap_count(joint)) { - struct ast_str *usbuf = ast_str_alloca(256); - struct ast_str *thembuf = ast_str_alloca(256); + struct ast_str *usbuf = ast_str_alloca(64); + struct ast_str *thembuf = ast_str_alloca(64); ast_rtp_codecs_payloads_destroy(&codecs); ast_log(LOG_NOTICE, "No joint capabilities for '%s' media stream between our configuration(%s) and incoming SDP(%s)\n", @@ -257,17 +257,36 @@ static int set_caps(struct ast_sip_session *session, struct ast_sip_session_medi ast_format_cap_append_from_cap(session->req_caps, joint, AST_MEDIA_TYPE_UNKNOWN); if (session->channel) { + struct ast_format *fmt; ast_channel_lock(session->channel); + ast_format_cap_remove_by_type(caps, AST_MEDIA_TYPE_UNKNOWN); + 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); + + /* + * XXX Historically we picked the "best" joint format to use + * and stuck with it. It would be nice to just append the + * determined joint media capabilities to give translation + * more formats to choose from when necessary. Unfortunately, + * there are some areas of the system where this doesn't work + * very well. (The softmix bridge in particular is reluctant + * to pick higher fidelity formats and has a problem with + * asymmetric sample rates.) + */ + fmt = ast_format_cap_get_format(joint, 0); + ast_format_cap_append(caps, fmt, 0); /* * Apply the new formats to the channel, potentially changing * raw read/write formats and translation path while doing so. */ - ast_channel_nativeformats_set(session->channel, joint); + ast_channel_nativeformats_set(session->channel, caps); ast_set_read_format(session->channel, ast_channel_readformat(session->channel)); ast_set_write_format(session->channel, ast_channel_writeformat(session->channel)); ast_channel_unlock(session->channel); + + ao2_ref(fmt, -1); } ast_rtp_codecs_payloads_destroy(&codecs); |