[asterisk-commits] rmudgett: branch 13 r423561 - in /branches/13: channels/ res/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Sep 19 12:08:51 CDT 2014
Author: rmudgett
Date: Fri Sep 19 12:08:47 2014
New Revision: 423561
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=423561
Log:
res_pjsip_sdp_rtp.c: Fix native formats containing formats that were not negotiated.
Outgoing PJSIP calls can result in non-negotiated formats listed in the
channel's native formats if video formats are listed in the endpoint's
configuration. The resulting call could then use a non-negotiated format
resulting in one way audio.
* Simplified the update of session->req_caps in set_caps(). Why do
something in five steps when only one is needed?
AFS-162 #close
Review: https://reviewboard.asterisk.org/r/4000/
Modified:
branches/13/channels/chan_pjsip.c
branches/13/res/res_pjsip_sdp_rtp.c
Modified: branches/13/channels/chan_pjsip.c
URL: http://svnview.digium.com/svn/asterisk/branches/13/channels/chan_pjsip.c?view=diff&rev=423561&r1=423560&r2=423561
==============================================================================
--- branches/13/channels/chan_pjsip.c (original)
+++ branches/13/channels/chan_pjsip.c Fri Sep 19 12:08:47 2014
@@ -417,6 +417,11 @@
}
ast_channel_nativeformats_set(chan, caps);
+
+ /*
+ * XXX Probably should pick the first audio codec instead
+ * of simply the first codec. The first codec may be video.
+ */
fmt = ast_format_cap_get_format(caps, 0);
ast_channel_set_writeformat(chan, fmt);
ast_channel_set_rawwriteformat(chan, fmt);
Modified: branches/13/res/res_pjsip_sdp_rtp.c
URL: http://svnview.digium.com/svn/asterisk/branches/13/res/res_pjsip_sdp_rtp.c?view=diff&rev=423561&r1=423560&r2=423561
==============================================================================
--- branches/13/res/res_pjsip_sdp_rtp.c (original)
+++ branches/13/res/res_pjsip_sdp_rtp.c Fri Sep 19 12:08:47 2014
@@ -256,18 +256,26 @@
ast_rtp_codecs_payloads_copy(&codecs, ast_rtp_instance_get_codecs(session_media->rtp),
session_media->rtp);
- ast_format_cap_remove_by_type(caps, AST_MEDIA_TYPE_UNKNOWN);
- ast_format_cap_append_from_cap(caps, session->req_caps, AST_MEDIA_TYPE_UNKNOWN);
- ast_format_cap_remove_by_type(caps, media_type);
- ast_format_cap_append_from_cap(caps, joint, AST_MEDIA_TYPE_UNKNOWN);
- ast_format_cap_append_from_cap(session->req_caps, caps, AST_MEDIA_TYPE_UNKNOWN);
+ 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);
More information about the asterisk-commits
mailing list