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asterisk
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Add SIP video fixes

Browse Source 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6448 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1.2-netsec
Mark Spencer 20 years ago
parent 538e160956
commit 5d4d2fdfd4
8 changed files with 78 additions and 3 deletions
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2
app.c
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@ -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

10
apps/app_dial.c
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@ -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))

3
apps/app_record.c
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@ -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;

5
channel.c
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@ -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;

45
channels/chan_sip.c
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@ -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)) {

1
file.c
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@ -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:

2
include/asterisk/frame.h
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@ -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)

13
rtp.c
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@ -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) ||

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