Thu Feb 20 07:00:00 CET 2003

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@619 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/asterisk.c b/asterisk.c
index e403c7f..5aedc34 100755 (executable)
--- a/asterisk.c
+++ b/asterisk.c
@@ -395,6 +395,7 @@ static void quit_handler(int num, int nice, int safeshutdown, int restart)
                ast_verbose("Asterisk %s ending (%d).\n", ast_active_channels() ? "uncleanly" : "cleanly", num);
        else if (option_debug)
                ast_log(LOG_DEBUG, "Asterisk ending (%d).\n", num);
+       manager_event(EVENT_FLAG_SYSTEM, "Shutdown", "Shutdown: %s\r\nRestart: %s\r\n", ast_active_channels() ? "Uncleanly" : "Cleanly", restart ? "True" : "False");
        if (ast_socket > -1) {
                close(ast_socket);
                ast_socket = -1;

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index f579f73..557e83b 100755 (executable)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -1632,8 +1632,10 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p, struct ast_rtp *
 {
        int len;
        int codec;
+       int alreadysent = 0;
        char costr[80];
        struct sockaddr_in sin;
+       struct sip_codec_pref *cur;
        char v[256];
        char s[256];
        char o[256];
@@ -1665,8 +1667,25 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p, struct ast_rtp *
        snprintf(c, sizeof(c), "c=IN IP4 %s\r\n", inet_ntoa(dest.sin_addr));
        snprintf(t, sizeof(t), "t=0 0\r\n");
        snprintf(m, sizeof(m), "m=audio %d RTP/AVP", ntohs(dest.sin_port));
+       /* Start by sending our preferred codecs */
+       cur = prefs;
+       while(cur) {
+               if (p->capability & cur->codec) {
+                       if (sipdebug)
+                               ast_verbose("Answering with preferred capability %d\n", cur->codec);
+                       if ((codec = ast2rtp(cur->codec)) > -1) {
+                               snprintf(costr, sizeof(costr), " %d", codec);
+                               strcat(m, costr);
+                               snprintf(costr, sizeof(costr), "a=rtpmap:%d %s/8000\r\n", codec, ast2rtpn(x));
+                               strcat(a, costr);
+                       }
+               }
+               alreadysent |= cur->codec;
+               cur = cur->next;
+       }
+       /* Now send anything else in no particular order */
        for (x=1;x<= AST_FORMAT_MAX_AUDIO; x <<= 1) {
-               if (p->capability & x) {
+               if ((p->capability & x) && !(alreadysent & x)) {
                        if (sipdebug)
                                ast_verbose("Answering with capability %d\n", x);
                        if ((codec = ast2rtp(x)) > -1) {

diff --git a/rtp.c b/rtp.c
index a93823c..aba47db 100755 (executable)
--- a/rtp.c
+++ b/rtp.c
@@ -592,11 +592,12 @@ static int ast_rtp_raw_write(struct ast_rtp *rtp, struct ast_frame *f, int codec
 
        /* Re-calculate last TS */
        rtp->lastts = ms * 8;
-       
+#if 0  /* XXX Experiment -- Make timestamp always relative XXX */
        /* If it's close to ou prediction, go for it */
        if (abs(rtp->lastts - pred) < 640)
+#endif 
                rtp->lastts = pred;
-#if 1
+#if 0
        else
                printf("Difference is %d, ms is %d\n", abs(rtp->lastts - pred), ms);
 #endif