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Merge in RTP-level packet bridging. Packet comes in, packet goes out - that's what RTP-level packet bridging is all about!

Browse Source 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41235 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1.4
Joshua Colp 19 years ago
parent d5572fac38
commit c70ed7614a
5 changed files with 444 additions and 239 deletions
Show Stats Download Patch File Download Diff File

18
channels/chan_mgcp.c
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@ -3876,13 +3876,19 @@ static struct mgcp_gateway *build_gateway(char *cat, struct ast_variable *v)
return (gw_reload ? NULL : gw);
}
static struct ast_rtp *mgcp_get_rtp_peer(struct ast_channel *chan)
static enum ast_rtp_get_result mgcp_get_rtp_peer(struct ast_channel *chan, struct ast_rtp **rtp)
{
struct mgcp_subchannel *sub;
sub = chan->tech_pvt;
if (sub && sub->rtp && sub->parent->canreinvite)
return sub->rtp;
return NULL;
struct mgcp_subchannel *sub = NULL;
if (!(sub = chan->tech_pvt) || !(sub->rtp))
return AST_RTP_GET_FAILED;
*rtp = sub->rtp;
if (sub->parent->canreinvite)
return AST_RTP_TRY_NATIVE;
else
return AST_RTP_TRY_PARTIAL;
}
static int mgcp_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, int codecs, int nat_active)

59
channels/chan_sip.c
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@ -1479,8 +1479,8 @@ static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_
/*----- RTP interface functions */
static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, int codecs, int nat_active);
static struct ast_rtp *sip_get_rtp_peer(struct ast_channel *chan);
static struct ast_rtp *sip_get_vrtp_peer(struct ast_channel *chan);
static enum ast_rtp_get_result sip_get_rtp_peer(struct ast_channel *chan, struct ast_rtp **rtp);
static enum ast_rtp_get_result sip_get_vrtp_peer(struct ast_channel *chan, struct ast_rtp **rtp);
static int sip_get_codec(struct ast_channel *chan);
static struct ast_frame *sip_rtp_read(struct ast_channel *ast, struct sip_pvt *p, int *faxdetect);
@ -16051,34 +16051,53 @@ static int sip_handle_t38_reinvite(struct ast_channel *chan, struct sip_pvt *pvt
/*! \brief Returns null if we can't reinvite audio (part of RTP interface) */
static struct ast_rtp *sip_get_rtp_peer(struct ast_channel *chan)
static enum ast_rtp_get_result sip_get_rtp_peer(struct ast_channel *chan, struct ast_rtp **rtp)
{
struct sip_pvt *p;
struct ast_rtp *rtp = NULL;
p = chan->tech_pvt;
if (!p)
return NULL;
struct sip_pvt *p = NULL;
enum ast_rtp_get_result res = AST_RTP_TRY_PARTIAL;
if (!(p = chan->tech_pvt))
return AST_RTP_GET_FAILED;
ast_mutex_lock(&p->lock);
if (p->rtp && ast_test_flag(&p->flags[0], SIP_CAN_REINVITE))
rtp = p->rtp;
if (!(p->rtp)) {
ast_mutex_unlock(&p->lock);
return AST_RTP_GET_FAILED;
}
*rtp = p->rtp;
if (ast_test_flag(&p->flags[0], SIP_CAN_REINVITE))
res = AST_RTP_TRY_NATIVE;
ast_mutex_unlock(&p->lock);
return rtp;
return res;
}
/*! \brief Returns null if we can't reinvite video (part of RTP interface) */
static struct ast_rtp *sip_get_vrtp_peer(struct ast_channel *chan)
static enum ast_rtp_get_result sip_get_vrtp_peer(struct ast_channel *chan, struct ast_rtp **rtp)
{
struct sip_pvt *p;
struct ast_rtp *rtp = NULL;
p = chan->tech_pvt;
if (!p)
return NULL;
struct sip_pvt *p = NULL;
enum ast_rtp_get_result res = AST_RTP_TRY_PARTIAL;
if (!(p = chan->tech_pvt))
return AST_RTP_GET_FAILED;
ast_mutex_lock(&p->lock);
if (p->vrtp && ast_test_flag(&p->flags[0], SIP_CAN_REINVITE))
rtp = p->vrtp;
if (!(p->rtp)) {
ast_mutex_unlock(&p->lock);
return AST_RTP_GET_FAILED;
}
*rtp = p->vrtp;
if (ast_test_flag(&p->flags[0], SIP_CAN_REINVITE))
res = AST_RTP_TRY_NATIVE;
ast_mutex_unlock(&p->lock);
return rtp;
return res;
}
/*! \brief Set the RTP peer for this call */

32
channels/chan_skinny.c
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@ -1591,24 +1591,28 @@ static void do_housekeeping(struct skinnysession *s)
/* I do not believe skinny can deal with video.
Anyone know differently? */
/* Yes, it can. Currently 7985 and Cisco VT Advantage do video. */
static struct ast_rtp *skinny_get_vrtp_peer(struct ast_channel *c)
static enum ast_rtp_get_result skinny_get_vrtp_peer(struct ast_channel *c, struct ast_rtp **rtp)
{
struct skinny_subchannel *sub;
sub = c->tech_pvt;
if (sub && sub->vrtp) {
return sub->vrtp;
}
return NULL;
struct skinny_subchannel *sub = NULL;
if (!(sub = c->tech_pvt) || !(sub->vrtp))
return AST_RTP_GET_FAILED;
*rtp = sub->vrtp;
return AST_RTP_TRY_NATIVE;
}
static struct ast_rtp *skinny_get_rtp_peer(struct ast_channel *c)
static enum ast_rtp_get_result skinny_get_rtp_peer(struct ast_channel *c, struct ast_rtp **rtp)
{
struct skinny_subchannel *sub;
sub = c->tech_pvt;
if (sub && sub->rtp) {
return sub->rtp;
}
return NULL;
struct skinny_subchannel *sub = NULL;
if (!(sub = c->tech_pvt) || !(sub->rtp))
return AST_RTP_GET_FAILED;
*rtp = sub->rtp;
return AST_RTP_TRY_NATIVE;
}
static int skinny_set_rtp_peer(struct ast_channel *c, struct ast_rtp *rtp, struct ast_rtp *vrtp, int codecs, int nat_active)

15
include/asterisk/rtp.h
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@ -54,11 +54,22 @@ enum ast_rtp_options {
AST_RTP_OPT_G726_NONSTANDARD = (1 << 0),
};
enum ast_rtp_get_result {
/*! Failed to find the RTP structure */
AST_RTP_GET_FAILED = 0,
/*! RTP structure exists but true native bridge can not occur so try partial */
AST_RTP_TRY_PARTIAL,
/*! RTP structure exists and native bridge can occur */
AST_RTP_TRY_NATIVE,
};
struct ast_rtp;
struct ast_rtp_protocol {
/*! Get RTP struct, or NULL if unwilling to transfer */
struct ast_rtp *(* const get_rtp_info)(struct ast_channel *chan);
enum ast_rtp_get_result (* const get_rtp_info)(struct ast_channel *chan, struct ast_rtp **rtp);
/*! Get RTP struct, or NULL if unwilling to transfer */
struct ast_rtp *(* const get_vrtp_info)(struct ast_channel *chan);
enum ast_rtp_get_result (* const get_vrtp_info)(struct ast_channel *chan, struct ast_rtp **rtp);
/*! Set RTP peer */
int (* const set_rtp_peer)(struct ast_channel *chan, struct ast_rtp *peer, struct ast_rtp *vpeer, int codecs, int nat_active);
int (* const get_codec)(struct ast_channel *chan);

559
main/rtp.c
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@ -155,6 +155,7 @@ struct ast_rtp {
int rtp_lookup_code_cache_code;
int rtp_lookup_code_cache_result;
struct ast_rtcp *rtcp;
struct ast_rtp *bridged; /*!< Who we are Packet briged to */
};
/* Forward declarations */
@ -169,6 +170,8 @@ static unsigned int ast_rtcp_calc_interval(struct ast_rtp *rtp);
#define FLAG_NAT_INACTIVE (0 << 1)
#define FLAG_NAT_INACTIVE_NOWARN (1 << 1)
#define FLAG_HAS_DTMF (1 << 3)
#define FLAG_P2P_SENT_MARK (1 << 4)
#define FLAG_P2P_NEED_DTMF (1 << 5)
/*!
* \brief Structure defining an RTCP session.
@ -926,6 +929,63 @@ static void calc_rxstamp(struct timeval *tv, struct ast_rtp *rtp, unsigned int t
rtp->rtcp->minrxjitter = rtp->rxjitter;
}
/*! \brief Perform a Packet2Packet write */
static int bridge_p2p_write(struct ast_rtp *rtp, unsigned int *rtpheader, int len, int hdrlen)
{
struct ast_rtp *bridged = rtp->bridged;
int res = 0, payload = 0, bridged_payload = 0, version, padding, mark, ext;
struct rtpPayloadType rtpPT;
unsigned int seqno;
/* Get fields from packet */
seqno = ntohl(rtpheader[0]);
version = (seqno & 0xC0000000) >> 30;
payload = (seqno & 0x7f0000) >> 16;
padding = seqno & (1 << 29);
mark = seqno & (1 << 23);
ext = seqno & (1 << 28);
seqno &= 0xffff;
/* Check what the payload value should be */
rtpPT = ast_rtp_lookup_pt(rtp, payload);
/* If the payload is DTMF, and we are listening for DTMF - then feed it into the core */
if (!rtpPT.isAstFormat && rtpPT.code == AST_RTP_DTMF)
return -1;
/* Otherwise adjust bridged payload to match */
bridged_payload = ast_rtp_lookup_code(bridged, rtpPT.isAstFormat, rtpPT.code);
/* If the mark bit has not been sent yet... do it now */
if (!ast_test_flag(rtp, FLAG_P2P_SENT_MARK)) {
mark = 1;
ast_set_flag(rtp, FLAG_P2P_SENT_MARK);
}
/* Reconstruct part of the packet */
rtpheader[0] = htonl((version << 30) | (mark << 23) | (bridged_payload << 16) | (seqno));
if (bridged->them.sin_port && bridged->them.sin_addr.s_addr) {
res = sendto(bridged->s, (void *)rtpheader, len, 0, (struct sockaddr *)&bridged->them, sizeof(bridged->them));
if (res < 0) {
if (!bridged->nat || (bridged->nat && (ast_test_flag(bridged, FLAG_NAT_ACTIVE) == FLAG_NAT_ACTIVE))) {
ast_log(LOG_DEBUG, "RTP Transmission error of packet to %s:%d: %s\n", ast_inet_ntoa(bridged->them.sin_addr), ntohs(bridged->them.sin_port), strerror(errno));
} else if ((ast_test_flag(bridged, FLAG_NAT_ACTIVE) == FLAG_NAT_INACTIVE) || rtpdebug) {
if (option_debug || rtpdebug)
ast_log(LOG_DEBUG, "RTP NAT: Can't write RTP to private address %s:%d, waiting for other end to send audio...\n", ast_inet_ntoa(bridged->them.sin_addr), ntohs(bridged->them.sin_port));
ast_set_flag(bridged, FLAG_NAT_INACTIVE_NOWARN);
}
return -1;
} else {
if (rtp_debug_test_addr(&bridged->them))
ast_verbose("Sent RTP P2P packet to %s:%d (type %-2.2d, len %-6.6u)\n", ast_inet_ntoa(bridged->them.sin_addr), ntohs(bridged->them.sin_port), bridged_payload, len - hdrlen);
return 0;
}
}
return -1;
}
struct ast_frame *ast_rtp_read(struct ast_rtp *rtp)
{
int res;
@ -964,6 +1024,11 @@ struct ast_frame *ast_rtp_read(struct ast_rtp *rtp)
return &ast_null_frame;
}
/* If we are P2P bridged to another channel, and the write is a success - then return a null frame and not the actual data */
if (rtp->bridged && !bridge_p2p_write(rtp, rtpheader, res, hdrlen)) {
return &ast_null_frame;
}
/* Get fields */
seqno = ntohl(rtpheader[0]);
@ -1285,9 +1350,11 @@ static struct ast_rtp_protocol *get_proto(struct ast_channel *chan)
int ast_rtp_early_bridge(struct ast_channel *dest, struct ast_channel *src)
{
struct ast_rtp *destp, *srcp=NULL; /* Audio RTP Channels */
struct ast_rtp *vdestp, *vsrcp=NULL; /* Video RTP channels */
struct ast_rtp_protocol *destpr, *srcpr=NULL;
struct ast_rtp *destp = NULL, *srcp = NULL; /* Audio RTP Channels */
struct ast_rtp *vdestp = NULL, *vsrcp = NULL; /* Video RTP channels */
struct ast_rtp_protocol *destpr = NULL, *srcpr = NULL;
enum ast_rtp_get_result audio_dest_res = AST_RTP_GET_FAILED, video_dest_res = AST_RTP_GET_FAILED;
enum ast_rtp_get_result audio_src_res = AST_RTP_GET_FAILED, video_src_res = AST_RTP_GET_FAILED;
int srccodec;
/* Lock channels */
@ -1322,27 +1389,27 @@ int ast_rtp_early_bridge(struct ast_channel *dest, struct ast_channel *src)
}
/* Get audio and video interface (if native bridge is possible) */
destp = destpr->get_rtp_info(dest);
vdestp = (destpr->get_vrtp_info) ? destpr->get_vrtp_info(dest) : NULL;
audio_dest_res = destpr->get_rtp_info(dest, &destp);
video_dest_res = destpr->get_vrtp_info ? destpr->get_vrtp_info(dest, &vdestp) : AST_RTP_GET_FAILED;
if (srcpr) {
srcp = srcpr->get_rtp_info(src);
vsrcp = (srcpr->get_vrtp_info) ? srcpr->get_vrtp_info(src) : NULL;
audio_src_res = srcpr->get_rtp_info(src, &srcp);
video_src_res = srcpr->get_vrtp_info ? srcpr->get_vrtp_info(src, &vsrcp) : AST_RTP_GET_FAILED;
}
/* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */
if (!destp) {
if (audio_dest_res != AST_RTP_TRY_NATIVE) {
/* Somebody doesn't want to play... */
ast_channel_unlock(dest);
if (src)
ast_channel_unlock(src);
return 0;
}
if (srcpr && srcpr->get_codec)
if (audio_src_res == AST_RTP_TRY_NATIVE && srcpr->get_codec)
srccodec = srcpr->get_codec(src);
else
srccodec = 0;
/* Consider empty media as non-existant */
if (srcp && !srcp->them.sin_addr.s_addr)
if (audio_src_res == AST_RTP_TRY_NATIVE && !srcp->them.sin_addr.s_addr)
srcp = NULL;
/* Bridge media early */
if (destpr->set_rtp_peer(dest, srcp, vsrcp, srccodec, srcp ? ast_test_flag(srcp, FLAG_NAT_ACTIVE) : 0))
@ -1357,10 +1424,13 @@ int ast_rtp_early_bridge(struct ast_channel *dest, struct ast_channel *src)
int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src, int media)
{
struct ast_rtp *destp, *srcp; /* Audio RTP Channels */
struct ast_rtp *vdestp, *vsrcp; /* Video RTP channels */
struct ast_rtp_protocol *destpr, *srcpr;
struct ast_rtp *destp = NULL, *srcp = NULL; /* Audio RTP Channels */
struct ast_rtp *vdestp = NULL, *vsrcp = NULL; /* Video RTP channels */
struct ast_rtp_protocol *destpr = NULL, *srcpr = NULL;
enum ast_rtp_get_result audio_dest_res = AST_RTP_GET_FAILED, video_dest_res = AST_RTP_GET_FAILED;
enum ast_rtp_get_result audio_src_res = AST_RTP_GET_FAILED, video_src_res = AST_RTP_GET_FAILED;
int srccodec;
/* Lock channels */
ast_channel_lock(dest);
while(ast_channel_trylock(src)) {
@ -1370,16 +1440,14 @@ int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src, i
}
/* Find channel driver interfaces */
destpr = get_proto(dest);
srcpr = get_proto(src);
if (!destpr) {
if (!(destpr = get_proto(dest))) {
if (option_debug)
ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", dest->name);
ast_channel_unlock(dest);
ast_channel_unlock(src);
return 0;
}
if (!srcpr) {
if (!(srcpr = get_proto(src))) {
if (option_debug)
ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", src->name);
ast_channel_unlock(dest);
@ -1388,13 +1456,13 @@ int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src, i
}
/* Get audio and video interface (if native bridge is possible) */
destp = destpr->get_rtp_info(dest);
vdestp = (destpr->get_vrtp_info) ? destpr->get_vrtp_info(dest) : NULL;
srcp = srcpr->get_rtp_info(src);
vsrcp = (srcpr->get_vrtp_info) ? srcpr->get_vrtp_info(src) : NULL;
audio_dest_res = destpr->get_rtp_info(dest, &destp);
video_dest_res = destpr->get_vrtp_info ? destpr->get_vrtp_info(dest, &vdestp) : AST_RTP_GET_FAILED;
audio_src_res = srcpr->get_rtp_info(src, &srcp);
video_src_res = srcpr->get_vrtp_info ? srcpr->get_vrtp_info(src, &vsrcp) : AST_RTP_GET_FAILED;
/* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */
if (!destp || !srcp) {
if (audio_dest_res != AST_RTP_TRY_NATIVE || audio_src_res != AST_RTP_TRY_NATIVE) {
/* Somebody doesn't want to play... */
ast_channel_unlock(dest);
ast_channel_unlock(src);
@ -1766,6 +1834,8 @@ void ast_rtp_stop(struct ast_rtp *rtp)
memset(&rtp->rtcp->them.sin_addr, 0, sizeof(rtp->rtcp->them.sin_addr));
memset(&rtp->rtcp->them.sin_port, 0, sizeof(rtp->rtcp->them.sin_port));
}
ast_clear_flag(rtp, FLAG_P2P_SENT_MARK);
}
void ast_rtp_reset(struct ast_rtp *rtp)
@ -2479,182 +2549,92 @@ int ast_rtp_proto_register(struct ast_rtp_protocol *proto)
return 0;
}
/*! \brief Bridge calls. If possible and allowed, initiate
re-invite so the peers exchange media directly outside
of Asterisk. */
enum ast_bridge_result ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, struct ast_frame **fo, struct ast_channel **rc, int timeoutms)
/*! \brief Bridge loop for true native bridge (reinvite) */
static enum ast_bridge_result bridge_native_loop(struct ast_channel *c0, struct ast_channel *c1, struct ast_rtp *p0, struct ast_rtp *p1, struct ast_rtp *vp0, struct ast_rtp *vp1, struct ast_rtp_protocol *pr0, struct ast_rtp_protocol *pr1, int codec0, int codec1, int timeoutms, int flags, struct ast_frame **fo, struct ast_channel **rc, void *pvt0, void *pvt1)
{
struct ast_frame *f;
struct ast_channel *who, *other, *cs[3];
struct ast_rtp *p0, *p1; /* Audio RTP Channels */
struct ast_rtp *vp0, *vp1; /* Video RTP channels */
struct ast_rtp_protocol *pr0, *pr1;
struct sockaddr_in ac0, ac1;
struct sockaddr_in vac0, vac1;
struct sockaddr_in t0, t1;
struct sockaddr_in vt0, vt1;
void *pvt0, *pvt1;
int codec0,codec1, oldcodec0, oldcodec1;
memset(&vt0, 0, sizeof(vt0));
memset(&vt1, 0, sizeof(vt1));
memset(&vac0, 0, sizeof(vac0));
memset(&vac1, 0, sizeof(vac1));
/* Lock channels */
ast_channel_lock(c0);
while(ast_channel_trylock(c1)) {
ast_channel_unlock(c0);
usleep(1);
ast_channel_lock(c0);
}
/* Find channel driver interfaces */
pr0 = get_proto(c0);
pr1 = get_proto(c1);
if (!pr0) {
ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", c0->name);
ast_channel_unlock(c0);
ast_channel_unlock(c1);
return AST_BRIDGE_FAILED;
}
if (!pr1) {
ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", c1->name);
ast_channel_unlock(c0);
ast_channel_unlock(c1);
return AST_BRIDGE_FAILED;
}
/* Get channel specific interface structures */
pvt0 = c0->tech_pvt;
pvt1 = c1->tech_pvt;
/* Get audio and video interface (if native bridge is possible) */
p0 = pr0->get_rtp_info(c0);
vp0 = pr0->get_vrtp_info ? pr0->get_vrtp_info(c0) : NULL;
p1 = pr1->get_rtp_info(c1);
vp1 = pr1->get_vrtp_info ? pr1->get_vrtp_info(c1) : NULL;
/* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */
if (!p0 || !p1) {
/* Somebody doesn't want to play... */
ast_channel_unlock(c0);
ast_channel_unlock(c1);
return AST_BRIDGE_FAILED_NOWARN;
}
if (ast_test_flag(p0, FLAG_HAS_DTMF) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) {
/* can't bridge, we are carrying DTMF for this channel and the bridge
needs it
*/
ast_channel_unlock(c0);
ast_channel_unlock(c1);
return AST_BRIDGE_FAILED_NOWARN;
}
if (ast_test_flag(p1, FLAG_HAS_DTMF) && (flags & AST_BRIDGE_DTMF_CHANNEL_1)) {
/* can't bridge, we are carrying DTMF for this channel and the bridge
needs it
*/
ast_channel_unlock(c0);
ast_channel_unlock(c1);
return AST_BRIDGE_FAILED_NOWARN;
}
/* Get codecs from both sides */
codec0 = pr0->get_codec ? pr0->get_codec(c0) : 0;
codec1 = pr1->get_codec ? pr1->get_codec(c1) : 0;
if (pr0->get_codec && pr1->get_codec) {
/* Hey, we can't do reinvite if both parties speak different codecs */
if (!(codec0 & codec1)) {
if (option_debug)
ast_log(LOG_DEBUG, "Channel codec0 = %d is not codec1 = %d, cannot native bridge in RTP.\n", codec0, codec1);
ast_channel_unlock(c0);
ast_channel_unlock(c1);
return AST_BRIDGE_FAILED_NOWARN;
}
}
struct ast_frame *fr = NULL;
struct ast_channel *who = NULL, *other = NULL, *cs[3] = {NULL, };
int oldcodec0 = codec0, oldcodec1 = codec1;
struct sockaddr_in ac1 = {0,}, vac1 = {0,}, ac0 = {0,}, vac0 = {0,};
struct sockaddr_in t1 = {0,}, vt1 = {0,}, t0 = {0,}, vt0 = {0,};
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "Native bridging %s and %s\n", c0->name, c1->name);
/* Set it up so audio goes directly between the two endpoints */
/* Ok, we should be able to redirect the media. Start with one channel */
if (pr0->set_rtp_peer(c0, p1, vp1, codec1, ast_test_flag(p1, FLAG_NAT_ACTIVE)))
ast_log(LOG_WARNING, "Channel '%s' failed to talk to '%s'\n", c0->name, c1->name);
else {
/* Store RTP peer */
/* Test the first channel */
if (!(pr0->set_rtp_peer(c0, p1, vp1, codec1, ast_test_flag(p1, FLAG_NAT_ACTIVE)))) {
ast_rtp_get_peer(p1, &ac1);
if (vp1)
ast_rtp_get_peer(vp1, &vac1);
}
/* Then test the other channel */
if (pr1->set_rtp_peer(c1, p0, vp0, codec0, ast_test_flag(p0, FLAG_NAT_ACTIVE)))
ast_log(LOG_WARNING, "Channel '%s' failed to talk back to '%s'\n", c1->name, c0->name);
else {
/* Store RTP peer */
} else
ast_log(LOG_WARNING, "Channel '%s' failed to talk to '%s'\n", c0->name, c1->name);
/* Test the second channel */
if (!(pr1->set_rtp_peer(c1, p0, vp0, codec0, ast_test_flag(p0, FLAG_NAT_ACTIVE)))) {
ast_rtp_get_peer(p0, &ac0);
if (vp0)
ast_rtp_get_peer(vp0, &vac0);
}
} else
ast_log(LOG_WARNING, "Channel '%s' failed to talk to '%s'\n", c1->name, c0->name);
/* Now we can unlock and move into our loop */
ast_channel_unlock(c0);
ast_channel_unlock(c1);
/* External RTP Bridge up, now loop and see if something happes that force us to take the
media back to Asterisk */
/* Throw our channels into the structure and enter the loop */
cs[0] = c0;
cs[1] = c1;
cs[2] = NULL;
oldcodec0 = codec0;
oldcodec1 = codec1;
for (;;) {
/* Check if something changed... */
if ((c0->tech_pvt != pvt0) ||
(c1->tech_pvt != pvt1) ||
(c0->masq || c0->masqr || c1->masq || c1->masqr)) {
ast_log(LOG_DEBUG, "Oooh, something is weird, backing out\n");
if (c0->tech_pvt == pvt0) {
if (pr0->set_rtp_peer(c0, NULL, NULL, 0, 0))
ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c0->name);
}
if (c1->tech_pvt == pvt1) {
if (pr1->set_rtp_peer(c1, NULL, NULL, 0, 0))
ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c1->name);
}
return AST_BRIDGE_RETRY;
/* Check if anything changed */
if ((c0->tech_pvt != pvt0) ||
(c1->tech_pvt != pvt1) ||
(c0->masq || c0->masqr || c1->masq || c1->masqr)) {
ast_log(LOG_DEBUG, "Oooh, something is weird, backing out\n");
if (c0->tech_pvt == pvt0)
if (pr0->set_rtp_peer(c0, NULL, NULL, 0, 0))
ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c0->name);
if (c1->tech_pvt == pvt1)
if (pr1->set_rtp_peer(c1, NULL, NULL, 0, 0))
ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c1->name);
return AST_BRIDGE_RETRY;
}
/* Now check if they have changed address */
/* Check if they have changed their address */
ast_rtp_get_peer(p1, &t1);
ast_rtp_get_peer(p0, &t0);
if (pr0->get_codec)
codec0 = pr0->get_codec(c0);
if (pr1->get_codec)
codec1 = pr1->get_codec(c1);
if (vp1)
ast_rtp_get_peer(vp1, &vt1);
if (pr1->get_codec)
codec1 = pr1->get_codec(c1);
ast_rtp_get_peer(p0, &t0);
if (vp0)
ast_rtp_get_peer(vp0, &vt0);
if (inaddrcmp(&t1, &ac1) || (vp1 && inaddrcmp(&vt1, &vac1)) || (codec1 != oldcodec1)) {
if (pr0->get_codec)
codec0 = pr0->get_codec(c0);
if ((inaddrcmp(&t1, &ac1)) ||
(vp1 && inaddrcmp(&vt1, &vac1)) ||
(codec1 != oldcodec1)) {
if (option_debug > 1) {
ast_log(LOG_DEBUG, "Oooh, '%s' changed end address to %s:%d (format %d)\n",
ast_log(LOG_DEBUG, "Oooh, '%s' changed end address to %s:%d (format %d)\n",
c1->name, ast_inet_ntoa(t1.sin_addr), ntohs(t1.sin_port), codec1);
ast_log(LOG_DEBUG, "Oooh, '%s' changed end vaddress to %s:%d (format %d)\n",
ast_log(LOG_DEBUG, "Oooh, '%s' changed end vaddress to %s:%d (format %d)\n",
c1->name, ast_inet_ntoa(vt1.sin_addr), ntohs(vt1.sin_port), codec1);
ast_log(LOG_DEBUG, "Oooh, '%s' was %s:%d/(format %d)\n",
ast_log(LOG_DEBUG, "Oooh, '%s' was %s:%d/(format %d)\n",
c1->name, ast_inet_ntoa(ac1.sin_addr), ntohs(ac1.sin_port), oldcodec1);
ast_log(LOG_DEBUG, "Oooh, '%s' was %s:%d/(format %d)\n",
ast_log(LOG_DEBUG, "Oooh, '%s' was %s:%d/(format %d)\n",
c1->name, ast_inet_ntoa(vac1.sin_addr), ntohs(vac1.sin_port), oldcodec1);
}
if (pr0->set_rtp_peer(c0, t1.sin_addr.s_addr ? p1 : NULL, vt1.sin_addr.s_addr ? vp1 : NULL, codec1, ast_test_flag(p1, FLAG_NAT_ACTIVE)))
if (pr0->set_rtp_peer(c0, t1.sin_addr.s_addr ? p1 : NULL, vt1.sin_addr.s_addr ? vp1 : NULL, codec1, ast_test_flag(p1, FLAG_NAT_ACTIVE)))
ast_log(LOG_WARNING, "Channel '%s' failed to update to '%s'\n", c0->name, c1->name);
memcpy(&ac1, &t1, sizeof(ac1));
memcpy(&vac1, &vt1, sizeof(vac1));
oldcodec1 = codec1;
}
if (inaddrcmp(&t0, &ac0) || (vp0 && inaddrcmp(&vt0, &vac0))) {
if (option_debug) {
ast_log(LOG_DEBUG, "Oooh, '%s' changed end address to %s:%d (format %d)\n",
if ((inaddrcmp(&t0, &ac0)) ||
(vp0 && inaddrcmp(&vt0, &vac0))) {
if (option_debug > 1) {
ast_log(LOG_DEBUG, "Oooh, '%s' changed end address to %s:%d (format %d)\n",
c0->name, ast_inet_ntoa(t0.sin_addr), ntohs(t0.sin_port), codec0);
ast_log(LOG_DEBUG, "Oooh, '%s' was %s:%d/(format %d)\n",
ast_log(LOG_DEBUG, "Oooh, '%s' was %s:%d/(format %d)\n",
c0->name, ast_inet_ntoa(ac0.sin_addr), ntohs(ac0.sin_port), oldcodec0);
}
if (pr1->set_rtp_peer(c1, t0.sin_addr.s_addr ? p0 : NULL, vt0.sin_addr.s_addr ? vp0 : NULL, codec0, ast_test_flag(p0, FLAG_NAT_ACTIVE)))
@ -2663,65 +2643,250 @@ enum ast_bridge_result ast_rtp_bridge(struct ast_channel *c0, struct ast_channel
memcpy(&vac0, &vt0, sizeof(vac0));
oldcodec0 = codec0;
}
who = ast_waitfor_n(cs, 2, &timeoutms);
if (!who) {
if (!timeoutms)
/* Wait for frame to come in on the channels */
if (!(who = ast_waitfor_n(cs, 2, &timeoutms))) {
if (!timeoutms)
return AST_BRIDGE_RETRY;
if (option_debug)
ast_log(LOG_DEBUG, "Ooh, empty read...\n");
/* check for hangup / whentohangup */
if (ast_check_hangup(c0) || ast_check_hangup(c1))
break;
continue;
}
f = ast_read(who);
other = (who == c0) ? c1 : c0; /* the other channel */
if (!f || ((f->frametype == AST_FRAME_DTMF) &&
(((who == c0) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) ||
((who == c1) && (flags & AST_BRIDGE_DTMF_CHANNEL_1))))) {
/* breaking out of the bridge. */
*fo = f;
fr = ast_read(who);
other = (who == c0) ? c1 : c0;
if (!fr || ((fr->frametype == AST_FRAME_DTMF) &&
(((who == c0) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) ||
((who == c1) && (flags & AST_BRIDGE_DTMF_CHANNEL_1))))) {
/* Break out of bridge */
*fo = fr;
*rc = who;
if (option_debug)
ast_log(LOG_DEBUG, "Oooh, got a %s\n", f ? "digit" : "hangup");
if ((c0->tech_pvt == pvt0)) {
if (pr0->set_rtp_peer(c0, NULL, NULL, 0, 0))
ast_log(LOG_DEBUG, "Oooh, got a %s\n", fr ? "digit" : "hangup");
if (c0->tech_pvt == pvt0)
if (pr0->set_rtp_peer(c0, NULL, NULL, 0, 0))
ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c0->name);
}
if ((c1->tech_pvt == pvt1)) {
if (pr1->set_rtp_peer(c1, NULL, NULL, 0, 0))
if (c1->tech_pvt == pvt1)
if (pr1->set_rtp_peer(c1, NULL, NULL, 0, 0))
ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c1->name);
return AST_BRIDGE_COMPLETE;
} else if ((fr->frametype == AST_FRAME_CONTROL) && !(flags & AST_BRIDGE_IGNORE_SIGS)) {
if ((fr->subclass == AST_CONTROL_HOLD) ||
(fr->subclass == AST_CONTROL_UNHOLD) ||
(fr->subclass == AST_CONTROL_VIDUPDATE)) {
ast_indicate(other, fr->subclass);
ast_frfree(fr);
} else {
*fo = fr;
*rc = who;
ast_log(LOG_DEBUG, "Got a FRAME_CONTROL (%d) frame on channel %s\n", fr->subclass, who->name);
return AST_BRIDGE_COMPLETE;
}
} else {
if ((fr->frametype == AST_FRAME_DTMF) ||
(fr->frametype == AST_FRAME_VOICE) ||
(fr->frametype == AST_FRAME_VIDEO)) {
ast_write(other, fr);
}
ast_frfree(fr);
}
/* Swap priority */
cs[2] = cs[0];
cs[0] = cs[1];
cs[1] = cs[2];
}
return AST_BRIDGE_FAILED;
}
/*! \brief Bridge loop for partial native bridge (packet2packet) */
static enum ast_bridge_result bridge_p2p_loop(struct ast_channel *c0, struct ast_channel *c1, struct ast_rtp *p0, struct ast_rtp *p1, struct ast_rtp *vp0, struct ast_rtp *vp1, int timeoutms, int flags, struct ast_frame **fo, struct ast_channel **rc, void *pvt0, void *pvt1)
{
struct ast_frame *fr = NULL;
struct ast_channel *who = NULL, *other = NULL, *cs[3] = {NULL, };
/* Okay, setup each RTP structure to do P2P forwarding */
ast_clear_flag(p0, FLAG_P2P_SENT_MARK);
p0->bridged = p1;
ast_clear_flag(p1, FLAG_P2P_SENT_MARK);
p1->bridged = p0;
if (vp0) {
ast_clear_flag(vp0, FLAG_P2P_SENT_MARK);
vp0->bridged = vp1;
ast_clear_flag(vp1, FLAG_P2P_SENT_MARK);
vp1->bridged = vp0;
}
/* Now let go of the channel locks and be on our way */
ast_channel_unlock(c0);
ast_channel_unlock(c1);
/* Go into a loop forwarding frames until we don't need to anymore */
cs[0] = c0;
cs[1] = c1;
cs[2] = NULL;
for (;;) {
/* Check if anything changed */
if ((c0->tech_pvt != pvt0) ||
(c1->tech_pvt != pvt1) ||
(c0->masq || c0->masqr || c1->masq || c1->masqr)) {
ast_log(LOG_DEBUG, "Oooh, something is weird, backing out\n");
return AST_BRIDGE_RETRY;
}
/* Wait on a channel to feed us a frame */
if (!(who = ast_waitfor_n(cs, 2, &timeoutms))) {
if (!timeoutms)
return AST_BRIDGE_RETRY;
if (option_debug)
ast_log(LOG_NOTICE, "Ooh, empty read...\n");
if (ast_check_hangup(c0) || ast_check_hangup(c1))
break;
continue;
}
/* Read in frame from channel */
fr = ast_read(who);
other = (who == c0) ? c1 : c0;
/* Dependong on the frame we may need to break out of our bridge */
if (!fr || ((fr->frametype == AST_FRAME_DTMF) &&
((who == c0) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) |
((who == c1) && (flags & AST_BRIDGE_DTMF_CHANNEL_1)))) {
/* Record received frame and who */
*fo = fr;
*rc = who;
if (option_debug)
ast_log(LOG_DEBUG, "Oooh, got a %s\n", fr ? "digit" : "hangup");
/* Break out of the bridge */
p0->bridged = NULL;
p1->bridged = NULL;
if (vp0) {
vp0->bridged = NULL;
vp1->bridged = NULL;
}
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(other, f->subclass);
ast_frfree(f);
} else if ((fr->frametype == AST_FRAME_CONTROL) && !(flags & AST_BRIDGE_IGNORE_SIGS)) {
if ((fr->subclass == AST_CONTROL_HOLD) ||
(fr->subclass == AST_CONTROL_UNHOLD) ||
(fr->subclass == AST_CONTROL_VIDUPDATE)) {
ast_indicate(other, fr->subclass);
ast_frfree(fr);
} else {
*fo = f;
*fo = fr;
*rc = who;
ast_log(LOG_DEBUG, "Got a FRAME_CONTROL (%d) frame on channel %s\n", f->subclass, who->name);
ast_log(LOG_DEBUG, "Got a FRAME_CONTROL (%d) frame on channel %s\n", fr->subclass, who->name);
return AST_BRIDGE_COMPLETE;
}
} else {
if ((f->frametype == AST_FRAME_DTMF) ||
(f->frametype == AST_FRAME_VOICE) ||
(f->frametype == AST_FRAME_VIDEO)) {
/* Forward voice or DTMF frames if they happen upon us */
ast_write(other, f);
/* If this is a DTMF, voice, or video frame write it to the other channel */
if ((fr->frametype == AST_FRAME_DTMF) ||
(fr->frametype == AST_FRAME_VOICE) ||
(fr->frametype == AST_FRAME_VIDEO)) {
ast_write(other, fr);
}
ast_frfree(f);
ast_frfree(fr);
}
/* Swap priority not that it's a big deal at this point */
/* Swap priority */
cs[2] = cs[0];
cs[0] = cs[1];
cs[1] = cs[2];
}
return AST_BRIDGE_FAILED;
}
/*! \brief Bridge calls. If possible and allowed, initiate
re-invite so the peers exchange media directly outside
of Asterisk. */
enum ast_bridge_result ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, struct ast_frame **fo, struct ast_channel **rc, int timeoutms)
{
struct ast_rtp *p0 = NULL, *p1 = NULL; /* Audio RTP Channels */
struct ast_rtp *vp0 = NULL, *vp1 = NULL; /* Video RTP channels */
struct ast_rtp_protocol *pr0 = NULL, *pr1 = NULL;
enum ast_rtp_get_result audio_p0_res = AST_RTP_GET_FAILED, video_p0_res = AST_RTP_GET_FAILED;
enum ast_rtp_get_result audio_p1_res = AST_RTP_GET_FAILED, video_p1_res = AST_RTP_GET_FAILED;
enum ast_bridge_result res = AST_BRIDGE_FAILED;
int codec0 = 0, codec1 = 0;
void *pvt0 = NULL, *pvt1 = NULL;
/* Lock channels */
ast_channel_lock(c0);
while(ast_channel_trylock(c1)) {
ast_channel_unlock(c0);
usleep(1);
ast_channel_lock(c0);
}
/* Find channel driver interfaces */
if (!(pr0 = get_proto(c0))) {
ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", c0->name);
ast_channel_unlock(c0);
ast_channel_unlock(c1);
return AST_BRIDGE_FAILED;
}
if (!(pr1 = get_proto(c1))) {
ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", c1->name);
ast_channel_unlock(c0);
ast_channel_unlock(c1);
return AST_BRIDGE_FAILED;
}
/* Get channel specific interface structures */
pvt0 = c0->tech_pvt;
pvt1 = c1->tech_pvt;
/* Get audio and video interface (if native bridge is possible) */
audio_p0_res = pr0->get_rtp_info(c0, &p0);
video_p0_res = pr0->get_vrtp_info ? pr0->get_vrtp_info(c0, &vp0) : AST_RTP_GET_FAILED;
audio_p1_res = pr1->get_rtp_info(c1, &p1);
video_p1_res = pr1->get_vrtp_info ? pr1->get_vrtp_info(c1, &vp1) : AST_RTP_GET_FAILED;
/* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */
if (audio_p0_res == AST_RTP_GET_FAILED || audio_p1_res == AST_RTP_GET_FAILED) {
/* Somebody doesn't want to play... */
ast_channel_unlock(c0);
ast_channel_unlock(c1);
return AST_BRIDGE_FAILED_NOWARN;
}
/* If we need to feed DTMF frames into the core then only do a partial native bridge */
if (ast_test_flag(p0, FLAG_HAS_DTMF) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) {
ast_set_flag(p0, FLAG_P2P_NEED_DTMF);
audio_p0_res = AST_RTP_TRY_PARTIAL;
}
if (ast_test_flag(p1, FLAG_HAS_DTMF) && (flags & AST_BRIDGE_DTMF_CHANNEL_1)) {
ast_set_flag(p1, FLAG_P2P_NEED_DTMF);
audio_p1_res = AST_RTP_TRY_PARTIAL;
}
/* Get codecs from both sides */
codec0 = pr0->get_codec ? pr0->get_codec(c0) : 0;
codec1 = pr1->get_codec ? pr1->get_codec(c1) : 0;
if (pr0->get_codec && pr1->get_codec) {
/* Hey, we can't do native bridging if both parties speak different codecs */
if (!(codec0 & codec1)) {
if (option_debug)
ast_log(LOG_DEBUG, "Channel codec0 = %d is not codec1 = %d, cannot native bridge in RTP.\n", codec0, codec1);
ast_channel_unlock(c0);
ast_channel_unlock(c1);
return AST_BRIDGE_FAILED_NOWARN;
}
}
/* If either side can only do a partial bridge, then don't try for a true native bridge */
if (audio_p0_res == AST_RTP_TRY_PARTIAL || audio_p1_res == AST_RTP_TRY_PARTIAL) {
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "Packet2Packet bridging %s and %s\n", c0->name, c1->name);
res = bridge_p2p_loop(c0, c1, p0, p1, vp0, vp1, timeoutms, flags, fo, rc, pvt0, pvt1);
} else {
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3 "Native bridging %s and %s\n", c0->name, c1->name);
res = bridge_native_loop(c0, c1, p0, p1, vp0, vp1, pr0, pr1, codec0, codec1, timeoutms, flags, fo, rc, pvt0, pvt1);
}
return res;
}
static int rtp_do_debug_ip(int fd, int argc, char *argv[])
{
struct hostent *hp;

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