diff options
| -rw-r--r-- | channels/chan_gtalk.c | 9 | ||||
| -rw-r--r-- | include/asterisk/rtp.h | 3 | ||||
| -rw-r--r-- | main/rtp.c | 10 |
3 files changed, 16 insertions, 6 deletions
diff --git a/channels/chan_gtalk.c b/channels/chan_gtalk.c index 6b039a174..59f40cc61 100644 --- a/channels/chan_gtalk.c +++ b/channels/chan_gtalk.c @@ -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(>alk_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; diff --git a/include/asterisk/rtp.h b/include/asterisk/rtp.h index c332159ec..d7738b345 100644 --- a/include/asterisk/rtp.h +++ b/include/asterisk/rtp.h @@ -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); diff --git a/main/rtp.c b/main/rtp.c index 5754710e8..53ef9f1c0 100644 --- a/main/rtp.c +++ b/main/rtp.c @@ -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 */ |