SIP hold improvements (bug #4290)
authorKevin P. Fleming <kpfleming@digium.com>
Mon, 16 May 2005 16:16:41 +0000 (16:16 +0000)
committerKevin P. Fleming <kpfleming@digium.com>
Mon, 16 May 2005 16:16:41 +0000 (16:16 +0000)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@5702 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_sip.c

index 7e44068..3e3a4c6 100755 (executable)
@@ -2136,7 +2136,7 @@ static int sip_answer(struct ast_channel *ast)
        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 @@ static int sip_write(struct ast_channel *ast, struct ast_frame *frame)
                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 @@ static int sip_write(struct ast_channel *ast, struct ast_frame *frame)
                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 @@ static struct ast_frame *sip_rtp_read(struct ast_channel *ast, struct sip_pvt *p
        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 @@ static void parse(struct sip_request *req)
                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 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
                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 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
                        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 @@ static int reload_config(void)
        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 @@ static struct ast_rtp *sip_get_rtp_peer(struct ast_channel *chan)
        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 @@ static struct ast_rtp *sip_get_vrtp_peer(struct ast_channel *chan)
        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 @@ static int sip_sipredirect(struct sip_pvt *p, const char *dest)
        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;