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Add support for H.264 with SIP and recording
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7855 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
3
rtp.c
3
rtp.c
@@ -655,6 +655,7 @@ static struct {
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{{1, AST_FORMAT_H261}, "video", "H261"},
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{{1, AST_FORMAT_H263}, "video", "H263"},
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{{1, AST_FORMAT_H263_PLUS}, "video", "h263-1998"},
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{{1, AST_FORMAT_H264}, "video", "H264"},
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};
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/* Static (i.e., well-known) RTP payload types for our "AST_FORMAT..."s:
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@@ -683,6 +684,7 @@ static struct rtpPayloadType static_RTP_PT[MAX_RTP_PT] = {
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[34] = {1, AST_FORMAT_H263},
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[103] = {1, AST_FORMAT_H263_PLUS},
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[97] = {1, AST_FORMAT_ILBC},
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[99] = {1, AST_FORMAT_H264},
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[101] = {0, AST_RTP_DTMF},
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[110] = {1, AST_FORMAT_SPEEX},
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[111] = {1, AST_FORMAT_G726},
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@@ -1515,6 +1517,7 @@ int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *_f)
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case AST_FORMAT_H261:
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case AST_FORMAT_H263:
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case AST_FORMAT_H263_PLUS:
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case AST_FORMAT_H264:
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case AST_FORMAT_G723_1:
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case AST_FORMAT_LPC10:
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case AST_FORMAT_SPEEX:
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