Update of /usr/cvsroot/asterisk/channels
In directory mongoose.digium.com:/tmp/cvs-serv15557/channels
Modified Files:
chan_sip.c
Log Message:
Add SIP video fixes
Index: chan_sip.c
===================================================================
RCS file: /usr/cvsroot/asterisk/channels/chan_sip.c,v
retrieving revision 1.826
retrieving revision 1.827
diff -u -d -r1.826 -r1.827
--- chan_sip.c 30 Aug 2005 01:04:17 -0000 1.826
+++ chan_sip.c 30 Aug 2005 02:12:09 -0000 1.827
@@ -832,6 +832,7 @@
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 @@
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 @@
/* 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 @@
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 @@
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 @@
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)) {
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