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Don't send empty RTP frames

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@788 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Mark Spencer
2003-04-08 21:53:46 +00:00
parent 6978d58552
commit 7f844be5e0
1 changed files with 5 additions and 0 deletions
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5
rtp.c
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@@ -750,12 +750,17 @@ int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *_f)
if (!rtp->them.sin_addr.s_addr) if (!rtp->them.sin_addr.s_addr)
return 0; return 0;
/* If there is no data length, return immediately */
if (!_f->datalen)
return 0;
/* Make sure we have enough space for RTP header */ /* Make sure we have enough space for RTP header */
if (_f->frametype != AST_FRAME_VOICE) { if (_f->frametype != AST_FRAME_VOICE) {
ast_log(LOG_WARNING, "RTP can only send voice\n"); ast_log(LOG_WARNING, "RTP can only send voice\n");
return -1; return -1;
} }
codec = ast_rtp_lookup_code(rtp, 1, _f->subclass); codec = ast_rtp_lookup_code(rtp, 1, _f->subclass);
if (codec < 0) { if (codec < 0) {
ast_log(LOG_WARNING, "Don't know how to send format %d packets with RTP\n", _f->subclass); ast_log(LOG_WARNING, "Don't know how to send format %d packets with RTP\n", _f->subclass);