diff options
| author | Richard Mudgett <rmudgett@digium.com> | 2017-12-11 18:20:06 -0600 |
|---|---|---|
| committer | Richard Mudgett <rmudgett@digium.com> | 2017-12-14 14:40:34 -0600 |
| commit | 98f7e9251f5dc9a7cf2eb82c77ec2796f00bce2f (patch) | |
| tree | 01ec0ae5189385aea305b1df76744c93cad6e632 /main/rtp_engine.c | |
| parent | 283d2df6809567c3663421e3a95a8ec19c670c6a (diff) | |
res_rtp_asterisk.c: Disable packet flood detection for video streams.
We should not do flood detection on video RTP streams. Video RTP streams
are very bursty by nature. They send out a burst of packets to update the
video frame then wait for the next video frame update. Really only audio
streams can be checked for flooding. The others are either bursty or
don't have a set rate.
* Added code to selectively disable packet flood detection for video RTP
streams.
ASTERISK-27440
Change-Id: I78031491a6e75c2d4b1e9c2462dc498fe9880a70
Diffstat (limited to 'main/rtp_engine.c')
| -rw-r--r-- | main/rtp_engine.c | 19 |
1 files changed, 19 insertions, 0 deletions
diff --git a/main/rtp_engine.c b/main/rtp_engine.c index 2431ffc0c..68c53e7ff 100644 --- a/main/rtp_engine.c +++ b/main/rtp_engine.c @@ -1176,6 +1176,25 @@ void ast_rtp_codecs_payloads_unset(struct ast_rtp_codecs *codecs, struct ast_rtp ast_rwlock_unlock(&codecs->codecs_lock); } +enum ast_media_type ast_rtp_codecs_get_stream_type(struct ast_rtp_codecs *codecs) +{ + enum ast_media_type stream_type = AST_MEDIA_TYPE_UNKNOWN; + int payload; + struct ast_rtp_payload_type *type; + + ast_rwlock_rdlock(&codecs->codecs_lock); + for (payload = 0; payload < AST_VECTOR_SIZE(&codecs->payload_mapping_rx); ++payload) { + type = AST_VECTOR_GET(&codecs->payload_mapping_rx, payload); + if (type && type->asterisk_format) { + stream_type = ast_format_get_type(type->format); + break; + } + } + ast_rwlock_unlock(&codecs->codecs_lock); + + return stream_type; +} + struct ast_rtp_payload_type *ast_rtp_codecs_get_payload(struct ast_rtp_codecs *codecs, int payload) { struct ast_rtp_payload_type *type = NULL; |