Improve RTP comments (bug #4792 with mods)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@6270 f38db490-d61c-443f-a65b-d21fe96a405b
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90f93deccc
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37
rtp.c
37
rtp.c
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@ -1370,6 +1370,7 @@ int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *_f)
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return 0;
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}
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/*--- ast_rtp_proto_unregister: Unregister interface to channel driver */
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void ast_rtp_proto_unregister(struct ast_rtp_protocol *proto)
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{
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struct ast_rtp_protocol *cur, *prev;
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@ -1389,6 +1390,7 @@ void ast_rtp_proto_unregister(struct ast_rtp_protocol *proto)
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}
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}
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/*--- ast_rtp_proto_register: Register interface to channel driver */
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int ast_rtp_proto_register(struct ast_rtp_protocol *proto)
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{
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struct ast_rtp_protocol *cur;
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@ -1405,9 +1407,11 @@ int ast_rtp_proto_register(struct ast_rtp_protocol *proto)
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return 0;
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}
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/*--- get_proto: Get channel driver interface structure */
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static struct ast_rtp_protocol *get_proto(struct ast_channel *chan)
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{
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struct ast_rtp_protocol *cur;
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cur = protos;
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while(cur) {
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if (cur->type == chan->type) {
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@ -1425,8 +1429,8 @@ int ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, st
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{
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struct ast_frame *f;
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struct ast_channel *who, *cs[3];
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struct ast_rtp *p0, *p1;
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struct ast_rtp *vp0, *vp1;
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struct ast_rtp *p0, *p1; /* Audio RTP Channels */
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struct ast_rtp *vp0, *vp1; /* Video RTP channels */
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struct ast_rtp_protocol *pr0, *pr1;
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struct sockaddr_in ac0, ac1;
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struct sockaddr_in vac0, vac1;
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@ -1446,12 +1450,16 @@ int ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, st
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/* if need DTMF, cant native bridge */
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if (flags & (AST_BRIDGE_DTMF_CHANNEL_0 | AST_BRIDGE_DTMF_CHANNEL_1))
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return -2;
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/* Lock channels */
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ast_mutex_lock(&c0->lock);
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while(ast_mutex_trylock(&c1->lock)) {
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ast_mutex_unlock(&c0->lock);
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usleep(1);
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ast_mutex_lock(&c0->lock);
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}
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/* Find channel driver interfaces */
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pr0 = get_proto(c0);
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pr1 = get_proto(c1);
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if (!pr0) {
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@ -1466,8 +1474,12 @@ int ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, st
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ast_mutex_unlock(&c1->lock);
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return -1;
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}
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/* Get channel specific interface structures */
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pvt0 = c0->tech_pvt;
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pvt1 = c1->tech_pvt;
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/* Get audio and video interface (if native bridge is possible) */
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p0 = pr0->get_rtp_info(c0);
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if (pr0->get_vrtp_info)
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vp0 = pr0->get_vrtp_info(c0);
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@ -1478,12 +1490,15 @@ int ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, st
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vp1 = pr1->get_vrtp_info(c1);
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else
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vp1 = NULL;
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/* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */
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if (!p0 || !p1) {
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/* Somebody doesn't want to play... */
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ast_mutex_unlock(&c0->lock);
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ast_mutex_unlock(&c1->lock);
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return -2;
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}
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/* Get codecs from both sides */
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if (pr0->get_codec)
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codec0 = pr0->get_codec(c0);
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else
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@ -1502,6 +1517,7 @@ int ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, st
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return -2;
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}
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}
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/* Ok, we should be able to redirect the media. Start with one channel */
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if (pr0->set_rtp_peer(c0, p1, vp1, codec1))
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ast_log(LOG_WARNING, "Channel '%s' failed to talk to '%s'\n", c0->name, c1->name);
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@ -1522,28 +1538,32 @@ int ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, st
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}
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ast_mutex_unlock(&c0->lock);
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ast_mutex_unlock(&c1->lock);
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/* External RTP Bridge up, now loop and see if something happes that force us to take the
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media back to Asterisk */
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cs[0] = c0;
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cs[1] = c1;
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cs[2] = NULL;
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oldcodec0 = codec0;
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oldcodec1 = codec1;
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for (;;) {
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/* Check if something changed... */
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if ((c0->tech_pvt != pvt0) ||
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(c1->tech_pvt != pvt1) ||
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(c0->masq || c0->masqr || c1->masq || c1->masqr)) {
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ast_log(LOG_DEBUG, "Oooh, something is weird, backing out\n");
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if (c0->tech_pvt == pvt0) {
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if (pr0->set_rtp_peer(c0, NULL, NULL, 0))
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ast_log(LOG_WARNING, "Channel '%s' failed to revert\n", c0->name);
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ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c0->name);
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}
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if (c1->tech_pvt == pvt1) {
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if (pr1->set_rtp_peer(c1, NULL, NULL, 0))
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ast_log(LOG_WARNING, "Channel '%s' failed to revert back\n", c1->name);
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ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c1->name);
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}
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/* Tell it to try again later */
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return -3;
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}
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to = -1;
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/* Now check if they have changed address */
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ast_rtp_get_peer(p1, &t1);
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ast_rtp_get_peer(p0, &t0);
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if (pr0->get_codec)
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@ -1555,7 +1575,7 @@ int ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, st
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if (vp0)
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ast_rtp_get_peer(vp0, &vt0);
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if (inaddrcmp(&t1, &ac1) || (vp1 && inaddrcmp(&vt1, &vac1)) || (codec1 != oldcodec1)) {
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if (option_debug) {
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if (option_debug > 1) {
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ast_log(LOG_DEBUG, "Oooh, '%s' changed end address to %s:%d (format %d)\n",
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c1->name, ast_inet_ntoa(iabuf, sizeof(iabuf), t1.sin_addr), ntohs(t1.sin_port), codec1);
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ast_log(LOG_DEBUG, "Oooh, '%s' changed end vaddress to %s:%d (format %d)\n",
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@ -1603,11 +1623,11 @@ int ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, st
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ast_log(LOG_DEBUG, "Oooh, got a %s\n", f ? "digit" : "hangup");
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if ((c0->tech_pvt == pvt0) && (!c0->_softhangup)) {
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if (pr0->set_rtp_peer(c0, NULL, NULL, 0))
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ast_log(LOG_WARNING, "Channel '%s' failed to revert\n", c0->name);
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ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c0->name);
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}
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if ((c1->tech_pvt == pvt1) && (!c1->_softhangup)) {
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if (pr1->set_rtp_peer(c1, NULL, NULL, 0))
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ast_log(LOG_WARNING, "Channel '%s' failed to revert back\n", c1->name);
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ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c1->name);
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}
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/* That's all we needed */
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return 0;
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@ -1666,7 +1686,7 @@ static int rtp_do_debug_ip(int fd, int argc, char *argv[])
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static int rtp_do_debug(int fd, int argc, char *argv[])
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{
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if(argc != 2){
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if(argc != 2) {
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if(argc != 4)
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return RESULT_SHOWUSAGE;
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return rtp_do_debug_ip(fd, argc, argv);
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@ -1749,6 +1769,7 @@ void ast_rtp_reload(void)
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}
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/*--- ast_rtp_init: Initialize the RTP system in Asterisk */
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void ast_rtp_init(void)
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{
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ast_cli_register(&cli_debug);
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