[asterisk-commits] branch oej/rtcp r32113 - in /team/oej/rtcp: ./
channels/
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Sun Jun 4 12:22:18 MST 2006
Author: oej
Date: Sun Jun 4 14:22:18 2006
New Revision: 32113
URL: http://svn.digium.com/view/asterisk?rev=32113&view=rev
Log:
Reset automerge
Modified:
team/oej/rtcp/ (props changed)
team/oej/rtcp/channel.c
team/oej/rtcp/channels/chan_sip.c
team/oej/rtcp/rtp.c
Propchange: team/oej/rtcp/
------------------------------------------------------------------------------
automerge = http://edvina.net/training/
Propchange: team/oej/rtcp/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Sun Jun 4 14:22:18 2006
@@ -1,1 +1,1 @@
-/trunk:1-32078
+/trunk:1-32111
Modified: team/oej/rtcp/channel.c
URL: http://svn.digium.com/view/asterisk/team/oej/rtcp/channel.c?rev=32113&r1=32112&r2=32113&view=diff
==============================================================================
--- team/oej/rtcp/channel.c (original)
+++ team/oej/rtcp/channel.c Sun Jun 4 14:22:18 2006
@@ -2097,7 +2097,7 @@
int ast_internal_timing_enabled(struct ast_channel *chan)
{
int ret = ast_opt_internal_timing && chan->timingfd > -1;
- if (option_debug > 20)
+ if (option_debug > 4)
ast_log(LOG_DEBUG, "Internal timing is %s (option_internal_timing=%d chan->timingfd=%d)\n", ret? "enabled": "disabled", ast_opt_internal_timing, chan->timingfd);
return ret;
}
Modified: team/oej/rtcp/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/oej/rtcp/channels/chan_sip.c?rev=32113&r1=32112&r2=32113&view=diff
==============================================================================
--- team/oej/rtcp/channels/chan_sip.c (original)
+++ team/oej/rtcp/channels/chan_sip.c Sun Jun 4 14:22:18 2006
@@ -805,7 +805,8 @@
int jointcapability; /*!< Supported capability at both ends (codecs ) */
int peercapability; /*!< Supported peer capability */
int prefcodec; /*!< Preferred codec (outbound only) */
- int noncodeccapability;
+ int noncodeccapability; /*!< DTMF RFC2833 telephony-event */
+ int redircodecs; /*!< Redirect codecs */
int maxcallbitrate; /*!< Maximum Call Bitrate for Video Calls */
int callingpres; /*!< Calling presentation */
int authtries; /*!< Times we've tried to authenticate */
@@ -817,7 +818,6 @@
struct sockaddr_in sa; /*!< Our peer */
struct sockaddr_in redirip; /*!< Where our RTP should be going if not to us */
struct sockaddr_in vredirip; /*!< Where our Video RTP should be going if not to us */
- int redircodecs; /*!< Redirect codecs */
struct sockaddr_in recv; /*!< Received as */
struct in_addr ourip; /*!< Our IP */
struct ast_channel *owner; /*!< Who owns us */
@@ -7329,12 +7329,8 @@
ast_uri_decode(referdata->replaces_callid);
if ((ptr = strchr(referdata->replaces_callid, ';'))) /* Remove options */ {
*ptr = '\0';
+ ptr++;
}
- /*
- * XXX don't know what was the intention but this code is
- * definitely wrong, as ptr can be NULL here.
- */
- ptr++;
/* Find the different tags before we destroy the string */
to = strcasestr(ptr, "to-tag=");
@@ -11223,7 +11219,8 @@
ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
return -1;
} else {
- /* XXX reduce nesting depth */
+ /* Save nesting depth for now, since there might be other events we will
+ support in the future */
/* Handle REFER notifications */
@@ -11323,13 +11320,6 @@
transmit_response(p, "200 OK", req);
return res;
};
-
- /* XXX hey, we never reach this code! */
- /* THis could be voicemail notification */
- transmit_response(p, "200 OK", req);
- if (!p->lastinvite)
- ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
- return res;
}
/*! \brief Handle incoming OPTIONS request */
@@ -11607,15 +11597,6 @@
/* Skip leading whitespace */
replace_id = ast_skip_blanks(replace_id);
- /* XXX there are several bugs in the code below,
- * because 'ptr' can be NULL so all the dereferences in strcasestr()
- * would cause panics.
- * I think we should do something like the code below, which also has
- * the advantage of not depending on the order of headers.
- * Please test if it works, and in case remove the block in #else / #endif
- */
-#if 1 /* proposed replacement */
-
start = replace_id;
while ( (ptr = strsep(&start, ";")) ) {
ptr = ast_skip_blanks(ptr); /* XXX maybe unnecessary ? */
@@ -11626,34 +11607,6 @@
fromtag = strsep(&fromtag, "&"); /* trim what ? */
}
}
-#else /* original code, buggy */
- if ((ptr = strchr(replace_id, ';'))) {
- *ptr = '\0';
- ptr++;
- }
- start = ptr;
-
- to = strcasestr(ptr, "to-tag=");
- if (to) {
- ptr = to + 7;
- totag = ptr;
- if ((to = strchr(ptr, ';')))
- *to = '\0';
- /* XXX this code is also wrong as to can be NULL */
- to++;
- ptr = to;
- }
-
- to = strcasestr(ptr, "from-tag=");
- if (to) {
- ptr = to + 9;
- fromtag = ptr;
- if ((to = strchr(ptr, '&')))
- *to = '\0';
- if ((to = strchr(ptr, ';')))
- *to = '\0';
- }
-#endif
if (sipdebug && option_debug > 3)
ast_log(LOG_DEBUG,"Invite/replaces: Will use Replace-Call-ID : %s Fromtag: %s Totag: %s\n", replace_id, fromtag ? fromtag : "<no from tag>", totag ? totag : "<no to tag>");
Modified: team/oej/rtcp/rtp.c
URL: http://svn.digium.com/view/asterisk/team/oej/rtcp/rtp.c?rev=32113&r1=32112&r2=32113&view=diff
==============================================================================
--- team/oej/rtcp/rtp.c (original)
+++ team/oej/rtcp/rtp.c Sun Jun 4 14:22:18 2006
@@ -1053,8 +1053,8 @@
rtp->themssrc = ntohl(rtpheader[2]); /* Record their SSRC to put in future RR */
if(rtp_debug_test_addr(&sin))
- ast_verbose("Got RTP packet from %s:%d (type %d, seq %d, ts %d, len %d)\n"
- , ast_inet_ntoa(iabuf, sizeof(iabuf), sin.sin_addr), ntohs(sin.sin_port), payloadtype, seqno, timestamp,res - hdrlen);
+ ast_verbose("Got RTP packet from %s:%d (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
+ ast_inet_ntoa(iabuf, sizeof(iabuf), sin.sin_addr), ntohs(sin.sin_port), payloadtype, seqno, timestamp,res - hdrlen);
rtpPT = ast_rtp_lookup_pt(rtp, payloadtype);
if (!rtpPT.isAstFormat) {
@@ -1890,7 +1890,7 @@
ast_inet_ntoa(iabuf, sizeof(iabuf), rtp->them.sin_addr),
ntohs(rtp->them.sin_port), strerror(errno));
if (rtp_debug_test_addr(&rtp->them))
- ast_verbose("Sent RTP packet to %s:%d (type %d, seq %u, ts %u, len %u)\n",
+ ast_verbose("Sent RTP DTMF packet to %s:%d (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
ast_inet_ntoa(iabuf, sizeof(iabuf), rtp->them.sin_addr),
ntohs(rtp->them.sin_port), payload, rtp->seqno, rtp->lastdigitts, res - hdrlen);
}
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