Add SIP video fixes
authorMark Spencer <markster@digium.com>
Tue, 30 Aug 2005 02:12:09 +0000 (02:12 +0000)
committerMark Spencer <markster@digium.com>
Tue, 30 Aug 2005 02:12:09 +0000 (02:12 +0000)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6448 65c4cc65-6c06-0410-ace0-fbb531ad65f3

app.c
apps/app_dial.c
apps/app_record.c
channel.c
channels/chan_sip.c
file.c
include/asterisk/frame.h
rtp.c

diff --git a/app.c b/app.c
index f3a84c3..4db4384 100755 (executable)
--- a/app.c
+++ b/app.c
@@ -615,6 +615,8 @@ int ast_play_and_record(struct ast_channel *chan, const char *playfile, const ch
                        return -1;
                }
        }
+       /* Request a video update */
+       ast_indicate(chan, AST_CONTROL_VIDUPDATE);
 
        if (x == fmtcnt) {
        /* Loop forever, writing the packets we read to the writer(s), until
index 8fe9fc4..ca72060 100755 (executable)
@@ -493,6 +493,11 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in, struct localu
                                                        if (!ast_test_flag(outgoing, DIAL_RINGBACKONLY))
                                                                ast_indicate(in, AST_CONTROL_PROGRESS);
                                                        break;
+                                               case AST_CONTROL_VIDUPDATE:
+                                                       if (option_verbose > 2)
+                                                               ast_verbose ( VERBOSE_PREFIX_3 "%s requested a video update, passing it to %s\n", o->chan->name,in->name);
+                                                       ast_indicate(in, AST_CONTROL_VIDUPDATE);
+                                                       break;
                                                case AST_CONTROL_PROCEEDING:
                                                        if (option_verbose > 2)
                                                                ast_verbose ( VERBOSE_PREFIX_3 "%s is proceeding passing it to %s\n", o->chan->name,in->name);
@@ -600,6 +605,11 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in, struct localu
                                if (ast_write(outgoing->chan, f))
                                        ast_log(LOG_WARNING, "Unable to forward voice\n");
                        }
+                       if (single && (f->frametype == AST_FRAME_CONTROL) && (f->subclass == AST_CONTROL_VIDUPDATE)) {
+                               if (option_verbose > 2)
+                                       ast_verbose ( VERBOSE_PREFIX_3 "%s requested a video update, passing it to %s\n", in->name,outgoing->chan->name);
+                               ast_indicate(outgoing->chan, AST_CONTROL_VIDUPDATE);
+                       }
                        ast_frfree(f);
                }
                if (!*to && (option_verbose > 2))
index c5da8f9..bb0e1ed 100755 (executable)
@@ -218,6 +218,9 @@ static int record_exec(struct ast_channel *chan, void *data)
                
                
                if (s) {
+                       /* Request a video update */
+                       ast_indicate(chan, AST_CONTROL_VIDUPDATE);
+
                        if (maxduration > 0)
                                timeout = time(NULL) + (time_t)maxduration;
                        
index 6fd536b..282a796 100755 (executable)
--- a/channel.c
+++ b/channel.c
@@ -1721,6 +1721,8 @@ int ast_indicate(struct ast_channel *chan, int condition)
                                /* Do nothing.... */
                        } else if (condition == AST_CONTROL_UNHOLD) {
                                /* Do nothing.... */
+                       } else if (condition == AST_CONTROL_VIDUPDATE) {
+                               /* Do nothing.... */
                        } else {
                                /* not handled */
                                ast_log(LOG_WARNING, "Unable to handle indication %d for '%s'\n", condition, chan->name);
@@ -2966,7 +2968,8 @@ static enum ast_bridge_result ast_generic_bridge(int *playitagain, int *playit,
                }
 
                if ((f->frametype == AST_FRAME_CONTROL) && !(config->flags & AST_BRIDGE_IGNORE_SIGS)) {
-                       if ((f->subclass == AST_CONTROL_HOLD) || (f->subclass == AST_CONTROL_UNHOLD)) {
+                       if ((f->subclass == AST_CONTROL_HOLD) || (f->subclass == AST_CONTROL_UNHOLD) ||
+                           (f->subclass == AST_CONTROL_VIDUPDATE)) {
                                ast_indicate(who == c0 ? c1 : c0, f->subclass);
                        } else {
                                *fo = f;
index 0b77fab..bdd2c36 100755 (executable)
@@ -832,6 +832,7 @@ static int transmit_request_with_auth(struct sip_pvt *p, int sipmethod, int inc,
 static int transmit_invite(struct sip_pvt *p, int sipmethod, int sendsdp, struct sip_invite_param *options, int init);
 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_info_with_vidupdate(struct sip_pvt *p);
 static int transmit_message_with_text(struct sip_pvt *p, const char *text);
 static int transmit_refer(struct sip_pvt *p, const char *dest);
 static int sip_sipredirect(struct sip_pvt *p, const char *dest);
@@ -2609,6 +2610,13 @@ static int sip_indicate(struct ast_channel *ast, int condition)
                        ast_log(LOG_DEBUG, "Bridged channel is back from hold, let's talk! : %s\n", p->callid);
                res = -1;
                break;
+       case AST_CONTROL_VIDUPDATE:     /* Request a video frame update */
+               if (p->vrtp && !ast_test_flag(p, SIP_NOVIDEO)) {
+                       transmit_info_with_vidupdate(p);
+                       res = 0;
+               } else
+                       res = -1;
+               break;
        case -1:
                res = -1;
                break;
@@ -3949,7 +3957,7 @@ static int __transmit_response(struct sip_pvt *p, char *msg, struct sip_request
        /* If we are cancelling an incoming invite for some reason, add information
                about the reason why we are doing this in clear text */
        if (p->owner && p->owner->hangupcause) {
-               add_header(&resp, "X-Asterisk-HangupCause:", ast_cause2str(p->owner->hangupcause));
+               add_header(&resp, "X-Asterisk-HangupCause", ast_cause2str(p->owner->hangupcause));
        }
        add_blank_header(&resp);
        return send_response(p, &resp, reliable, seqno);
@@ -4056,6 +4064,26 @@ static int add_digit(struct sip_request *req, char digit)
        return 0;
 }
 
+/*--- add_vidupdate: add XML encoded media control with update ---*/
+/* XML: The only way to turn 0 bits of information into a few hundred. */
+static int add_vidupdate(struct sip_request *req)
+{
+       const char *xml_is_a_huge_waste_of_space =
+               "<?xml version=\"1.0\" encoding=\"utf-8\" ?>\r\n"
+               " <media_control>\r\n"
+               "  <vc_primitive>\r\n"
+               "   <to_encoder>\r\n"
+               "    <picture_fast_update\r\n"
+               "    </picture_fast_update>\r\n"
+               "   </to_encoder>\r\n"
+               "  </vc_primitive>\r\n"
+               " </media_control>\r\n";
+       add_header(req, "Content-Type", "application/media_control+xml");
+       add_header_contentLength(req, strlen(xml_is_a_huge_waste_of_space));
+       add_line(req, xml_is_a_huge_waste_of_space);
+       return 0;
+}
+
 /*--- add_sdp: Add Session Description Protocol message ---*/
 static int add_sdp(struct sip_request *resp, struct sip_pvt *p)
 {
@@ -5209,6 +5237,15 @@ static int transmit_info_with_digit(struct sip_pvt *p, char digit)
        return send_request(p, &req, 1, p->ocseq);
 }
 
+/*--- transmit_info_with_vidupdate: Send SIP INFO with video update request ---*/
+static int transmit_info_with_vidupdate(struct sip_pvt *p)
+{
+       struct sip_request req;
+       reqprep(&req, p, SIP_INFO, 0, 1);
+       add_vidupdate(&req);
+       return send_request(p, &req, 1, p->ocseq);
+}
+
 /*--- transmit_request: transmit generic SIP request ---*/
 static int transmit_request(struct sip_pvt *p, int sipmethod, int seqno, int reliable, int newbranch)
 {
@@ -8125,6 +8162,12 @@ static void handle_request_info(struct sip_pvt *p, struct sip_request *req)
                        ast_set_flag(p, SIP_NEEDDESTROY);
                }
                return;
+       } else if (!strcasecmp(get_header(req, "Content-Type"), "application/media_control+xml")) {
+               /* Eh, we'll just assume it's a fast picture update for now */
+               if (p->owner)
+                       ast_queue_control(p->owner, AST_CONTROL_VIDUPDATE);
+               transmit_response(p, "200 OK", req);
+               return;
        } else if ((c = get_header(req, "X-ClientCode"))) {
                /* Client code (from SNOM phone) */
                if (ast_test_flag(p, SIP_USECLIENTCODE)) {
diff --git a/file.c b/file.c
index 6fcd311..ab24719 100755 (executable)
--- a/file.c
+++ b/file.c
@@ -987,6 +987,7 @@ int ast_waitstream(struct ast_channel *c, const char *breakon)
                                        return -1;
                                case AST_CONTROL_RINGING:
                                case AST_CONTROL_ANSWER:
+                               case AST_CONTROL_VIDUPDATE:
                                        /* Unimportant */
                                        break;
                                default:
index 41a4e7c..364aaa4 100755 (executable)
@@ -194,6 +194,8 @@ struct ast_frame_chain {
 #define AST_CONTROL_HOLD                       16
 /*! Indicate call is left from hold */
 #define AST_CONTROL_UNHOLD                     17
+/*! Indicate video frame update */
+#define AST_CONTROL_VIDUPDATE          18
 
 #define AST_SMOOTHER_FLAG_G729         (1 << 0)
 
diff --git a/rtp.c b/rtp.c
index 71a2436..e71b2e0 100755 (executable)
--- a/rtp.c
+++ b/rtp.c
@@ -672,7 +672,7 @@ void ast_rtp_pt_default(struct ast_rtp* rtp)
        rtp->rtp_lookup_code_cache_result = 0;
 }
 
-/* Make a note of a RTP payload type that was seen in a SDP "m=" line. */
+/* Make a note of a RTP paymoad type that was seen in a SDP "m=" line. */
 /* By default, use the well-known value for this type (although it may */
 /* still be set to a different value by a subsequent "a=rtpmap:" line): */
 void ast_rtp_set_m_type(struct ast_rtp* rtp, int pt) {
@@ -1628,6 +1628,17 @@ enum ast_bridge_result ast_rtp_bridge(struct ast_channel *c0, struct ast_channel
                                        ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c1->name);
                        }
                        return AST_BRIDGE_COMPLETE;
+               } else if ((f->frametype == AST_FRAME_CONTROL) && !(flags & AST_BRIDGE_IGNORE_SIGS)) {
+                       if ((f->subclass == AST_CONTROL_HOLD) || (f->subclass == AST_CONTROL_UNHOLD) ||
+                           (f->subclass == AST_CONTROL_VIDUPDATE)) {
+                               ast_indicate(who == c0 ? c1 : c0, f->subclass);
+                               ast_frfree(f);
+                       } else {
+                               *fo = f;
+                               *rc = who;
+                               ast_log(LOG_DEBUG, "Got a FRAME_CONTROL (%d) frame on channel %s\n", f->subclass, who->name);
+                               return AST_BRIDGE_COMPLETE;
+                       }
                } else {
                        if ((f->frametype == AST_FRAME_DTMF) || 
                                (f->frametype == AST_FRAME_VOICE) ||