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Make SIP early media work more efficiently without so many reinvites

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@26019 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Mark Spencer
2006-05-09 11:44:50 +00:00
parent 66dda468d1
commit 9953f4f40e
4 changed files with 123 additions and 15 deletions
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7
include/asterisk/rtp.h
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@@ -97,7 +97,8 @@ struct ast_rtp *ast_rtp_new_with_bindaddr(struct sched_context *sched, struct io
void ast_rtp_set_peer(struct ast_rtp *rtp, struct sockaddr_in *them);
void ast_rtp_get_peer(struct ast_rtp *rtp, struct sockaddr_in *them);
/* Copies from rtp to them and returns 1 if there was a change or 0 if it was already the same */
int ast_rtp_get_peer(struct ast_rtp *rtp, struct sockaddr_in *them);
void ast_rtp_get_us(struct ast_rtp *rtp, struct sockaddr_in *us);
@@ -154,7 +155,9 @@ 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);
int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src, int media);
int ast_rtp_early_media(struct ast_channel *dest, struct ast_channel *src);
void ast_rtp_stop(struct ast_rtp *rtp);