diff --git a/app.c b/app.c
index f3a84c3a59f4d76760337f430071d6dc42376dac..4db43846b3a63170e62adb7c3f1956c6547d4693 100755
--- 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
diff --git a/apps/app_dial.c b/apps/app_dial.c
index 8fe9fc43ad562ddcf7dbc1b92fef2426021e8382..ca72060b9ea9b26efbf763fa64007357bc3a1588 100755
--- a/apps/app_dial.c
+++ b/apps/app_dial.c
@@ -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))
diff --git a/apps/app_record.c b/apps/app_record.c
index c5da8f91027d7bfcd2f3446ed700b664a4506473..bb0e1ed0007a4ad2423847c8f40a6fb6872fb768 100755
--- a/apps/app_record.c
+++ b/apps/app_record.c
@@ -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;
diff --git a/channel.c b/channel.c
index 6fd536b7f0fe345e18c666e83d394f837a01962e..282a7967e90288c2bc33c6d17b77e26c96a893e2 100755
--- 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;
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 0b77fabc4a50dc1f43f7859eee7861ab954c4ae4..bdd2c36e88c1ab0b54af65f0765d896314d4924d 100755
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -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 6fcd31173f6d2074b6f896dd11954f50a4449059..ab24719cbb920268a8d7dcdc976de65f3559ac4d 100755
--- 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:
diff --git a/include/asterisk/frame.h b/include/asterisk/frame.h
index 41a4e7cb5f172f9c5ec8110d2a251947d38ecf21..364aaa4cc55d95750405bb9ce3116cbaa52758fb 100755
--- a/include/asterisk/frame.h
+++ b/include/asterisk/frame.h
@@ -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 71a24360f7892e5c9c46e07aebf79bb1eb41895c..e71b2e0223a6368c6663aad20e71265eb6218102 100755
--- 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) ||