[asterisk-commits] file: trunk r43340 - /trunk/main/rtp.c
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Wed Sep 20 09:55:09 MST 2006
Author: file
Date: Wed Sep 20 11:55:09 2006
New Revision: 43340
URL: http://svn.digium.com/view/asterisk?rev=43340&view=rev
Log:
Expand codec check so that raw formats must be equal for a Packet2Packet bridge to occur
Modified:
trunk/main/rtp.c
Modified: trunk/main/rtp.c
URL: http://svn.digium.com/view/asterisk/trunk/main/rtp.c?rev=43340&r1=43339&r2=43340&view=diff
==============================================================================
--- trunk/main/rtp.c (original)
+++ trunk/main/rtp.c Wed Sep 20 11:55:09 2006
@@ -3066,19 +3066,25 @@
/* Get codecs from both sides */
codec0 = pr0->get_codec ? pr0->get_codec(c0) : 0;
codec1 = pr1->get_codec ? pr1->get_codec(c1) : 0;
- if (pr0->get_codec && pr1->get_codec) {
+ if (codec0 && codec1 && !(codec0 & codec1)) {
/* Hey, we can't do native bridging if both parties speak different codecs */
- if (!(codec0 & codec1)) {
+ if (option_debug)
+ ast_log(LOG_DEBUG, "Channel codec0 = %d is not codec1 = %d, cannot native bridge in RTP.\n", codec0, codec1);
+ ast_channel_unlock(c0);
+ ast_channel_unlock(c1);
+ return AST_BRIDGE_FAILED_NOWARN;
+ }
+
+ /* If either side can only do a partial bridge, then don't try for a true native bridge */
+ if (audio_p0_res == AST_RTP_TRY_PARTIAL || audio_p1_res == AST_RTP_TRY_PARTIAL) {
+ /* In order to do Packet2Packet bridging both sides must be in the same rawread/rawwrite */
+ if (c0->rawreadformat != c1->rawwriteformat || c1->rawreadformat != c0->rawwriteformat) {
if (option_debug)
- ast_log(LOG_DEBUG, "Channel codec0 = %d is not codec1 = %d, cannot native bridge in RTP.\n", codec0, codec1);
+ ast_log(LOG_DEBUG, "Cannot packet2packet bridge - raw formats are incompatible\n");
ast_channel_unlock(c0);
ast_channel_unlock(c1);
return AST_BRIDGE_FAILED_NOWARN;
}
- }
-
- /* If either side can only do a partial bridge, then don't try for a true native bridge */
- if (audio_p0_res == AST_RTP_TRY_PARTIAL || audio_p1_res == AST_RTP_TRY_PARTIAL) {
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "Packet2Packet bridging %s and %s\n", c0->name, c1->name);
res = bridge_p2p_loop(c0, c1, p0, p1, vp0, vp1, timeoutms, flags, fo, rc, pvt0, pvt1);
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