Doc fix for controlplayback, get rid of 500ms wait in rtp.c (bug #1589)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@3090 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/apps/app_controlplayback.c b/apps/app_controlplayback.c
index 089a7f4..850ebed 100755 (executable)
--- a/apps/app_controlplayback.c
+++ b/apps/app_controlplayback.c
@@ -31,14 +31,13 @@ static char *app = "ControlPlayback";
 static char *synopsis = "Play a file with fast forward and rewind";
 
 static char *descrip = 
-"ControlPlayback(filename[|skipms]|[ffchar]|[rewchar]|[stopchar]]):\n"
+"ControlPlayback(filename[|skipms[|ffchar[|rewchar[|stopchar[|pausechr]]]]]):\n"
 "  Plays  back  a  given  filename (do not put extension). Options may also\n"
 "  be included following a pipe symbol.  You can use * and # to rewind and\n"
 "  fast forward the playback specified. If 'stopchar' is added the file will\n"
 "  terminate playback when 'stopchar' is pressed. Returns -1 if the channel\n"
 "  was hung up, or if the file does not exist. Returns 0 otherwise.\n\n"
-"  Example:  exten => 1234,1,ControlPlayback(file|4000|*|#|1)\n\n";
-
+"  Example:  exten => 1234,1,ControlPlayback(file|4000|*|#|1|0)\n\n";
 
 STANDARD_LOCAL_USER;
 

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index a9b75da..15c3f81 100755 (executable)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -265,6 +265,7 @@ static struct sip_pvt {
        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 */
@@ -305,6 +306,7 @@ static struct sip_pvt {
        char lastmsg[256];                      /* Last Message sent/received */
        int amaflags;                           /* AMA Flags */
        int pendinginvite;                      /* Any pending invite */
+       int needreinvite;                       /* Do we need to send another reinvite? */
        int pendingbye;                         /* Need to send bye after we ack? */
        int gotrefer;                           /* Got a refer? */
        struct sip_request initreq;             /* Initial request */
@@ -493,7 +495,7 @@ static int transmit_response_with_auth(struct sip_pvt *p, char *msg, struct sip_
 static int transmit_request(struct sip_pvt *p, char *msg, int inc, int reliable, int newbranch);
 static int transmit_request_with_auth(struct sip_pvt *p, char *msg, int inc, int reliable, int newbranch);
 static int transmit_invite(struct sip_pvt *p, char *msg, int sendsdp, char *auth, char *authheader, char *vxml_url,char *distinctive_ring, int init);
-static int transmit_reinvite_with_sdp(struct sip_pvt *p, struct ast_rtp *rtp, struct ast_rtp *vrtp, int codec);
+static int transmit_reinvite_with_sdp(struct sip_pvt *p);
 static int transmit_info_with_digit(struct sip_pvt *p, char digit);
 static int transmit_message_with_text(struct sip_pvt *p, char *text);
 static int transmit_refer(struct sip_pvt *p, char *dest);
@@ -1552,6 +1554,7 @@ static int sip_hangup(struct ast_channel *ast)
                                /* Note we will need a BYE when this all settles out
                                   but we can't send one while we have "INVITE" outstanding. */
                                p->pendingbye = 1;
+                               p->needreinvite = 0;
                        }
                }
        }
@@ -3003,7 +3006,7 @@ static int add_digit(struct sip_request *req, char digit)
 }
 
 /*--- add_sdp: Add Session Description Protocol message ---*/
-static int add_sdp(struct sip_request *resp, struct sip_pvt *p, struct ast_rtp *rtp, struct ast_rtp *vrtp, int codecs)
+static int add_sdp(struct sip_request *resp, struct sip_pvt *p)
 {
        int len;
        int codec;
@@ -3033,8 +3036,6 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p, struct ast_rtp *
                return -1;
        }
        capability = p->jointcapability;
-       if (codecs) 
-               capability = codecs & p->jointcapability;
                
        if (!p->sessionid) {
                p->sessionid = getpid();
@@ -3048,8 +3049,8 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p, struct ast_rtp *
        if (p->redirip.sin_addr.s_addr) {
                dest.sin_port = p->redirip.sin_port;
                dest.sin_addr = p->redirip.sin_addr;
-       } else if (rtp) {
-               ast_rtp_get_peer(rtp, &dest);
+               if (p->redircodecs)
+                       capability = p->redircodecs;
        } else {
                dest.sin_addr = p->ourip;
                dest.sin_port = sin.sin_port;
@@ -3060,8 +3061,6 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p, struct ast_rtp *
                if (p->vredirip.sin_addr.s_addr) {
                        vdest.sin_port = p->vredirip.sin_port;
                        vdest.sin_addr = p->vredirip.sin_addr;
-               } else if (vrtp) {
-                       ast_rtp_get_peer(vrtp, &vdest);
                } else {
                        vdest.sin_addr = p->ourip;
                        vdest.sin_port = vsin.sin_port;
@@ -3210,7 +3209,7 @@ static int transmit_response_with_sdp(struct sip_pvt *p, char *msg, struct sip_r
                return -1;
        }
        respprep(&resp, p, msg, req);
-       add_sdp(&resp, p, NULL, NULL, 0);
+       add_sdp(&resp, p);
        return send_response(p, &resp, retrans, seqno);
 }
 
@@ -3277,7 +3276,7 @@ static int determine_firstline_parts( struct sip_request *req ) {
 /* transmit_reinvite_with_sdp: Transmit reinvite with SDP :-) ---*/
 /*   A re-invite is basically a new INVITE with the same CALL-ID and TAG as the
      INVITE that opened the SIP dialogue */
-static int transmit_reinvite_with_sdp(struct sip_pvt *p, struct ast_rtp *rtp, struct ast_rtp *vrtp, int codec)
+static int transmit_reinvite_with_sdp(struct sip_pvt *p)
 {
        struct sip_request req;
        if (p->canreinvite == REINVITE_UPDATE)
@@ -3286,7 +3285,7 @@ static int transmit_reinvite_with_sdp(struct sip_pvt *p, struct ast_rtp *rtp, st
                reqprep(&req, p, "INVITE", 0, 1);
        
        add_header(&req, "Allow", ALLOWED_METHODS);
-       add_sdp(&req, p, rtp, vrtp, codec);
+       add_sdp(&req, p);
        /* Use this as the basis */
        copy_request(&p->initreq, &req);
        parse(&p->initreq);
@@ -3433,7 +3432,7 @@ static int transmit_invite(struct sip_pvt *p, char *cmd, int sdp, char *auth, ch
        }
        add_header(&req, "Allow", ALLOWED_METHODS);
        if (sdp) {
-               add_sdp(&req, p, NULL, NULL, 0);
+               add_sdp(&req, p);
        } else {
                add_header(&req, "Content-Length", "0");
                add_blank_header(&req);
@@ -5846,6 +5845,21 @@ static void parse_moved_contact(struct sip_pvt *p, struct sip_request *req)
                strncpy(p->owner->call_forward, s, sizeof(p->owner->call_forward) - 1);
 }
 
+static void check_pendings(struct sip_pvt *p)
+{
+       /* Go ahead and send bye at this point */
+       if (p->pendingbye) {
+               transmit_request_with_auth(p, "BYE", 0, 1, 1);
+               p->needdestroy = 1;
+               p->needreinvite = 0;
+       } else if (p->needreinvite) {
+               ast_log(LOG_DEBUG, "Sending pending reinvite on '%s'\n", p->callid);
+               /* Didn't get to reinvite yet, so do it now */
+               transmit_reinvite_with_sdp(p);
+               p->needreinvite = 0;
+       }
+}
+
 /*--- handle_response: Handle SIP response in dialogue ---*/
 static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int ignore)
 {
@@ -5984,11 +5998,7 @@ static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_
                                p->authtries = 0;
                                /* If I understand this right, the branch is different for a non-200 ACK only */
                                transmit_request(p, "ACK", seqno, 0, 1);
-                               /* Go ahead and send bye at this point */
-                               if (p->pendingbye) {
-                                       transmit_request_with_auth(p, "BYE", 0, 1, 1);
-                                       p->needdestroy = 1;
-                               }
+                               check_pendings(p);
                        } else if (!strcasecmp(msg, "REGISTER")) {
                                /* char *exp; */
                                int expires, expires_ms;
@@ -6342,6 +6352,8 @@ static int handle_request(struct sip_pvt *p, struct sip_request *req, struct soc
                        sip_cancel_destroy(p);
                        /* This call is no longer outgoing if it ever was */
                        p->outgoing = 0;
+                       /* This also counts as a pending invite */
+                       p->pendinginvite = 1;
                        copy_request(&p->initreq, req);
                        check_via(p, req);
                        if (!ast_strlen_zero(get_header(req, "Content-Type"))) {
@@ -6681,11 +6693,13 @@ static int handle_request(struct sip_pvt *p, struct sip_request *req, struct soc
                /* Uhm, I haven't figured out the point of the ACK yet.  Are we
                   supposed to retransmit responses until we get an ack? 
                   Make sure this is on a valid call */
+               p->pendinginvite = 0;
                __sip_ack(p, seqno, FLAG_RESPONSE);
                if (!ast_strlen_zero(get_header(req, "Content-Type"))) {
                        if (process_sdp(p, req))
                                return -1;
                } 
+               check_pendings(p);
                if (!p->lastinvite && ast_strlen_zero(p->randdata))
                        p->needdestroy = 1;
        } else if (!strcasecmp(cmd, "SIP/2.0")) {
@@ -7730,7 +7744,7 @@ static struct ast_rtp *sip_get_vrtp_peer(struct ast_channel *chan)
        return NULL;
 }
 
-static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, int codec)
+static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, int codecs)
 {
        struct sip_pvt *p;
        p = chan->pvt->pvt;
@@ -7743,8 +7757,14 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struc
                        ast_rtp_get_peer(vrtp, &p->vredirip);
                else
                        memset(&p->vredirip, 0, sizeof(p->vredirip));
+               p->redircodecs = codecs;
                if (!p->gotrefer) {
-                       transmit_reinvite_with_sdp(p, rtp, vrtp, codec);
+                       if (!p->pendinginvite)
+                               transmit_reinvite_with_sdp(p);
+                       else if (!p->pendingbye) {
+                               ast_log(LOG_DEBUG, "Deferring reinvite on '%s'\n", p->callid);
+                               p->needreinvite = 1;
+                       }
                        p->outgoing = 1;
                }
                return 0;

diff --git a/rtp.c b/rtp.c
index 12658cf..90c7d60 100755 (executable)
--- a/rtp.c
+++ b/rtp.c
@@ -1264,14 +1264,6 @@ int ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, st
        memset(&vac0, 0, sizeof(vac0));
        memset(&vac1, 0, sizeof(vac1));
 
-       /* XXX Wait a half a second for things to settle up 
-                       this really should be fixed XXX */
-       ast_autoservice_start(c0);
-       ast_autoservice_start(c1);
-       usleep(500000);
-       ast_autoservice_stop(c0);
-       ast_autoservice_stop(c1);
-
        /* if need DTMF, cant native bridge */
        if (flags & (AST_BRIDGE_DTMF_CHANNEL_0 | AST_BRIDGE_DTMF_CHANNEL_1))
                return -2;