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

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@3090 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1.0
Mark Spencer 21 years ago
parent cf57ba2310
commit b820fd0075
3 changed files with 41 additions and 30 deletions
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5
apps/app_controlplayback.c
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@ -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;

58
channels/chan_sip.c
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@ -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;

8
rtp.c
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@ -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;

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