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Do not do a partial bridge for Google Talk since we need to handle STUN. (issue #8448 reported by phsultan)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48168 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Joshua Colp
2006-11-30 21:18:24 +00:00
parent 76e35bac11
commit b2b70adede
3 changed files with 16 additions and 6 deletions
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9
channels/chan_gtalk.c
View File

@@ -163,7 +163,6 @@ struct gtalk_container {
};
static const char desc[] = "Gtalk Channel";
static const char type[] = "Gtalk";
static int usecnt = 0;
AST_MUTEX_DEFINE_STATIC(usecnt_lock);
@@ -195,7 +194,7 @@ static int gtalk_get_codec(struct ast_channel *chan);
/*! \brief PBX interface structure for channel registration */
static const struct ast_channel_tech gtalk_tech = {
.type = type,
.type = "Gtalk",
.description = "Gtalk Channel Driver",
.capabilities = ((AST_FORMAT_MAX_AUDIO << 1) - 1),
.requester = gtalk_request,
@@ -223,7 +222,7 @@ static struct in_addr __ourip;
/*! \brief RTP driver interface */
static struct ast_rtp_protocol gtalk_rtp = {
type: "gtalk",
type: "Gtalk",
get_rtp_info: gtalk_get_rtp_peer,
set_rtp_peer: gtalk_set_rtp_peer,
get_codec: gtalk_get_codec,
@@ -922,10 +921,12 @@ static struct ast_channel *gtalk_new(struct gtalk *client, struct gtalk_pvt *i,
fmt = ast_best_codec(tmp->nativeformats);
if (i->rtp) {
ast_rtp_setstun(i->rtp, 1);
tmp->fds[0] = ast_rtp_fd(i->rtp);
tmp->fds[1] = ast_rtcp_fd(i->rtp);
}
if (i->vrtp) {
ast_rtp_setstun(i->rtp, 1);
tmp->fds[2] = ast_rtp_fd(i->vrtp);
tmp->fds[3] = ast_rtcp_fd(i->vrtp);
}
@@ -1796,7 +1797,7 @@ static int load_module(void)
/* Make sure we can register our channel type */
if (ast_channel_register(&gtalk_tech)) {
ast_log(LOG_ERROR, "Unable to register channel class %s\n", type);
ast_log(LOG_ERROR, "Unable to register channel class %s\n", gtalk_tech.type);
return -1;
}
return 0;

3
include/asterisk/rtp.h
View File

@@ -186,6 +186,9 @@ void ast_rtp_setdtmf(struct ast_rtp *rtp, int dtmf);
/*! \brief Compensate for devices that send RFC2833 packets all at once */
void ast_rtp_setdtmfcompensate(struct ast_rtp *rtp, int compensate);
/*! \brief Enable STUN capability */
void ast_rtp_setstun(struct ast_rtp *rtp, int stun_enable);
int ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, struct ast_frame **fo, struct ast_channel **rc, int timeoutms);
int ast_rtp_proto_register(struct ast_rtp_protocol *proto);

10
main/rtp.c
View File

@@ -183,6 +183,7 @@ static int bridge_p2p_rtcp_write(struct ast_rtp *rtp, unsigned int *rtcpheader,
#define FLAG_P2P_NEED_DTMF (1 << 5)
#define FLAG_CALLBACK_MODE (1 << 6)
#define FLAG_DTMF_COMPENSATE (1 << 7)
#define FLAG_HAS_STUN (1 << 8)
/*!
* \brief Structure defining an RTCP session.
@@ -545,6 +546,11 @@ void ast_rtp_setdtmfcompensate(struct ast_rtp *rtp, int compensate)
ast_set2_flag(rtp, compensate ? 1 : 0, FLAG_DTMF_COMPENSATE);
}
void ast_rtp_setstun(struct ast_rtp *rtp, int stun_enable)
{
ast_set2_flag(rtp, stun_enable ? 1 : 0, FLAG_HAS_STUN);
}
static struct ast_frame *send_dtmf(struct ast_rtp *rtp, enum ast_frame_type type)
{
if (((ast_test_flag(rtp, FLAG_DTMF_COMPENSATE) && type == AST_FRAME_DTMF_END) ||
@@ -2843,8 +2849,8 @@ static int p2p_rtp_callback(int *id, int fd, short events, void *cbdata)
/*! \brief Helper function to switch a channel and RTP stream into callback mode */
static int p2p_callback_enable(struct ast_channel *chan, struct ast_rtp *rtp, int *fds, int **iod)
{
/* If we need DTMF or we have no IO structure, then we can't do direct callback */
if (ast_test_flag(rtp, FLAG_P2P_NEED_DTMF) || !rtp->io)
/* If we need DTMF, are looking for STUN, or we have no IO structure then we can't do direct callback */
if (ast_test_flag(rtp, FLAG_P2P_NEED_DTMF) || ast_test_flag(rtp, FLAG_HAS_STUN) || !rtp->io)
return 0;
/* If the RTP structure is already in callback mode, remove it temporarily */