[Asterisk-cvs] asterisk/channels chan_sip.c,1.733,1.734
kpfleming at lists.digium.com
kpfleming at lists.digium.com
Mon May 16 12:11:22 CDT 2005
Update of /usr/cvsroot/asterisk/channels
In directory mongoose.digium.com:/tmp/cvs-serv24995/channels
Modified Files:
chan_sip.c
Log Message:
SIP hold improvements (bug #4290)
Index: chan_sip.c
===================================================================
RCS file: /usr/cvsroot/asterisk/channels/chan_sip.c,v
retrieving revision 1.733
retrieving revision 1.734
diff -u -d -r1.733 -r1.734
--- chan_sip.c 15 May 2005 04:17:42 -0000 1.733
+++ chan_sip.c 16 May 2005 16:16:41 -0000 1.734
@@ -2136,7 +2136,7 @@
return res;
}
-/*--- sip_write: Send response, support audio media ---*/
+/*--- sip_write: Send frame to media channel (rtp) ---*/
static int sip_write(struct ast_channel *ast, struct ast_frame *frame)
{
struct sip_pvt *p = ast->tech_pvt;
@@ -2151,6 +2151,7 @@
if (p) {
ast_mutex_lock(&p->lock);
if (p->rtp) {
+ /* If channel is not up, activate early media session */
if ((ast->_state != AST_STATE_UP) && !ast_test_flag(p, SIP_PROGRESS_SENT) && !ast_test_flag(p, SIP_OUTGOING)) {
transmit_response_with_sdp(p, "183 Session Progress", &p->initreq, 0);
ast_set_flag(p, SIP_PROGRESS_SENT);
@@ -2165,6 +2166,7 @@
if (p) {
ast_mutex_lock(&p->lock);
if (p->vrtp) {
+ /* Activate video early media */
if ((ast->_state != AST_STATE_UP) && !ast_test_flag(p, SIP_PROGRESS_SENT) && !ast_test_flag(p, SIP_OUTGOING)) {
transmit_response_with_sdp(p, "183 Session Progress", &p->initreq, 0);
ast_set_flag(p, SIP_PROGRESS_SENT);
@@ -2562,7 +2564,7 @@
default:
f = &null_frame;
}
- /* Don't send RFC2833 if we're not supposed to */
+ /* Don't forward RFC2833 if we're not supposed to */
if (f && (f->frametype == AST_FRAME_DTMF) && (ast_test_flag(p, SIP_DTMF) != SIP_DTMF_RFC2833))
return &null_frame;
if (p->owner) {
@@ -2953,7 +2955,7 @@
ast_log(LOG_WARNING, "Odd content, extra stuff left over ('%s')\n", c);
}
-/*--- process_sdp: Process SIP SDP ---*/
+/*--- process_sdp: Process SIP SDP and activate RTP channels---*/
static int process_sdp(struct sip_pvt *p, struct sip_request *req)
{
char *m;
@@ -3161,7 +3163,8 @@
ast_log(LOG_NOTICE, "No compatible codecs!\n");
return -1;
}
- if (p->owner) {
+ if (p->owner) { /* There's an open channel owning us */
+ struct ast_channel *bridgepeer = NULL;
if (!(p->owner->nativeformats & p->jointcapability)) {
const unsigned slen=512;
char s1[slen], s2[slen];
@@ -3172,28 +3175,49 @@
ast_set_read_format(p->owner, p->owner->readformat);
ast_set_write_format(p->owner, p->owner->writeformat);
}
- if (ast_bridged_channel(p->owner)) {
+ if ((bridgepeer=ast_bridged_channel(p->owner))) {
+ /* We have a bridge */
/* Turn on/off music on hold if we are holding/unholding */
if (sin.sin_addr.s_addr && !sendonly) {
- ast_moh_stop(ast_bridged_channel(p->owner));
+ ast_moh_stop(bridgepeer);
+ /* Indicate UNHOLD status to the other channel */
+ ast_indicate(bridgepeer, AST_CONTROL_UNHOLD);
+ append_history(p, "Unhold", req->data);
if (callevents && ast_test_flag(p, SIP_CALL_ONHOLD)) {
manager_event(EVENT_FLAG_CALL, "Unhold",
"Channel: %s\r\n"
"Uniqueid: %s\r\n",
p->owner->name,
p->owner->uniqueid);
- ast_clear_flag(p, SIP_CALL_ONHOLD);
}
+ ast_clear_flag(p, SIP_CALL_ONHOLD);
+ /* Somehow, we need to check if we need to re-invite here */
+ /* If this call had a native bridge, it's broken
+ now and we need to start all over again.
+ The bridged peer, if SIP, now listens
+ to RTP from Asterisk instead of from
+ the peer
+
+ So IF we had a native bridge before
+ the HOLD, we need to somehow re-invite
+ into a NATIVE bridge afterwards...
+
+ */
+
} else {
+ /* No address for RTP, we're on hold */
+ append_history(p, "Hold", req->data);
if (callevents && !ast_test_flag(p, SIP_CALL_ONHOLD)) {
manager_event(EVENT_FLAG_CALL, "Hold",
"Channel: %s\r\n"
"Uniqueid: %s\r\n",
p->owner->name,
p->owner->uniqueid);
- ast_set_flag(p, SIP_CALL_ONHOLD);
}
- ast_moh_start(ast_bridged_channel(p->owner), NULL);
+ ast_set_flag(p, SIP_CALL_ONHOLD);
+ /* Indicate HOLD status to the other channel */
+ ast_indicate(bridgepeer, AST_CONTROL_HOLD);
+ ast_moh_start(bridgepeer, NULL);
if (sendonly)
ast_rtp_stop(p->rtp);
}
@@ -10734,7 +10758,7 @@
return 0;
}
-/*--- sip_get_rtp_peer: Returns null if we can't reinvite */
+/*--- sip_get_rtp_peer: Returns null if we can't reinvite (part of RTP interface) */
static struct ast_rtp *sip_get_rtp_peer(struct ast_channel *chan)
{
struct sip_pvt *p;
@@ -10749,6 +10773,7 @@
return rtp;
}
+/*--- sip_get_vrtp_peer: Returns null if we can't reinvite video (part of RTP interface) */
static struct ast_rtp *sip_get_vrtp_peer(struct ast_channel *chan)
{
struct sip_pvt *p;
@@ -10764,7 +10789,8 @@
return rtp;
}
-/*--- sip_set_rtp_peer: Set the RTP peer for this call ---*/
+/*--- sip_set_rtp_peer: Set the data needed to RE-INVITE this call
+ so that the peers media go between them, outside of Asterisk. ---*/
static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, int codecs)
{
struct sip_pvt *p;
@@ -11042,7 +11068,7 @@
return -1;
}
-/*--- sip_get_codec: Return peers codec ---*/
+/*--- sip_get_codec: Return SIP UA's codec (part of the RTP interface) ---*/
static int sip_get_codec(struct ast_channel *chan)
{
struct sip_pvt *p = chan->tech_pvt;
More information about the svn-commits
mailing list