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Merged revisions 48168 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48169 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
10
main/rtp.c
10
main/rtp.c
@@ -183,6 +183,7 @@ static int bridge_p2p_rtcp_write(struct ast_rtp *rtp, unsigned int *rtcpheader,
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#define FLAG_P2P_NEED_DTMF (1 << 5)
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#define FLAG_CALLBACK_MODE (1 << 6)
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#define FLAG_DTMF_COMPENSATE (1 << 7)
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#define FLAG_HAS_STUN (1 << 8)
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/*!
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* \brief Structure defining an RTCP session.
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@@ -545,6 +546,11 @@ void ast_rtp_setdtmfcompensate(struct ast_rtp *rtp, int compensate)
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ast_set2_flag(rtp, compensate ? 1 : 0, FLAG_DTMF_COMPENSATE);
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}
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void ast_rtp_setstun(struct ast_rtp *rtp, int stun_enable)
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{
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ast_set2_flag(rtp, stun_enable ? 1 : 0, FLAG_HAS_STUN);
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}
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static struct ast_frame *send_dtmf(struct ast_rtp *rtp, enum ast_frame_type type)
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{
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if (((ast_test_flag(rtp, FLAG_DTMF_COMPENSATE) && type == AST_FRAME_DTMF_END) ||
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@@ -2913,8 +2919,8 @@ static int p2p_rtp_callback(int *id, int fd, short events, void *cbdata)
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/*! \brief Helper function to switch a channel and RTP stream into callback mode */
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static int p2p_callback_enable(struct ast_channel *chan, struct ast_rtp *rtp, int **iod)
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{
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/* If we need DTMF or we have no IO structure, then we can't do direct callback */
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if (ast_test_flag(rtp, FLAG_P2P_NEED_DTMF) || !rtp->io)
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/* If we need DTMF, are looking for STUN, or we have no IO structure then we can't do direct callback */
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if (ast_test_flag(rtp, FLAG_P2P_NEED_DTMF) || ast_test_flag(rtp, FLAG_HAS_STUN) || !rtp->io)
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return 0;
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/* If the RTP structure is already in callback mode, remove it temporarily */
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