Merged revisions 48168 via svnmerge from

https://origsvn.digium.com/svn/asterisk/branches/1.4

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r48168 | file | 2006-11-30 16:18:24 -0500 (Thu, 30 Nov 2006) | 2 lines

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/trunk@48169 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Joshua Colp
2006-11-30 21:22:01 +00:00
parent 813f8d26fe
commit 869101028b
3 changed files with 16 additions and 6 deletions

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) ||
@@ -2913,8 +2919,8 @@ 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 **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 */