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Major RTP fixes for using inbound SDP on outbound connection, get rid of

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old local rtp stuff...


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7551 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Mark Spencer
2005-12-20 17:52:31 +00:00
parent 93d1d0dcfb
commit 0d32a85be1
7 changed files with 151 additions and 47 deletions
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4
include/asterisk/frame.h
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@@ -452,6 +452,10 @@ extern int ast_codec_get_samples(struct ast_frame *f);
/*! \brief Returns the number of bytes for the number of samples of the given format */
extern int ast_codec_get_len(int format, int samples);
/*! \brief Appends a frame to the end of a list of frames, truncating the maximum length of the list */
extern struct ast_frame *ast_frame_enqueue(struct ast_frame *head, struct ast_frame *f, int maxlen, int dupe);
/*! \brief Gets duration in ms of interpolation frame for a format */
static inline int ast_codec_interp_len(int format)
{

3
include/asterisk/rtp.h
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@@ -135,7 +135,6 @@ void ast_rtp_set_rtpmap_type(struct ast_rtp* rtp, int pt,
/* Mapping between RTP payload format codes and Asterisk codes: */
struct rtpPayloadType ast_rtp_lookup_pt(struct ast_rtp* rtp, int pt);
int ast_rtp_lookup_code(struct ast_rtp* rtp, int isAstFormat, int code);
void ast_rtp_offered_from_local(struct ast_rtp* rtp, int local);
void ast_rtp_get_current_formats(struct ast_rtp* rtp,
int* astFormats, int* nonAstFormats);
@@ -154,6 +153,8 @@ int ast_rtp_proto_register(struct ast_rtp_protocol *proto);
void ast_rtp_proto_unregister(struct ast_rtp_protocol *proto);
int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src);
void ast_rtp_stop(struct ast_rtp *rtp);
void ast_rtp_init(void);