Don't send empty RTP frames
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@788 f38db490-d61c-443f-a65b-d21fe96a405b
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5
rtp.c
5
rtp.c
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@ -749,6 +749,10 @@ int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *_f)
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/* If we have no peer, return immediately */
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if (!rtp->them.sin_addr.s_addr)
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return 0;
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/* If there is no data length, return immediately */
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if (!_f->datalen)
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return 0;
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/* Make sure we have enough space for RTP header */
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if (_f->frametype != AST_FRAME_VOICE) {
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@ -756,6 +760,7 @@ int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *_f)
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return -1;
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}
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codec = ast_rtp_lookup_code(rtp, 1, _f->subclass);
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if (codec < 0) {
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ast_log(LOG_WARNING, "Don't know how to send format %d packets with RTP\n", _f->subclass);
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