diff options
| author | Richard Mudgett <rmudgett@digium.com> | 2017-12-11 18:20:06 -0600 |
|---|---|---|
| committer | Richard Mudgett <rmudgett@digium.com> | 2017-12-14 14:40:17 -0600 |
| commit | 61e81338d97da2b6c0bf52be3cd66e9017db4dc9 (patch) | |
| tree | fdac0eac9cec323482ba4fef9c9ab7b697eeaac4 /include | |
| parent | 9a016bd65e21235ea0827666e270974dd7faea0a (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 'include')
| -rw-r--r-- | include/asterisk/rtp_engine.h | 10 |
1 files changed, 10 insertions, 0 deletions
diff --git a/include/asterisk/rtp_engine.h b/include/asterisk/rtp_engine.h index 0b29f3485..7daff6738 100644 --- a/include/asterisk/rtp_engine.h +++ b/include/asterisk/rtp_engine.h @@ -1348,6 +1348,16 @@ int ast_rtp_codecs_payloads_set_rtpmap_type_rate(struct ast_rtp_codecs *codecs, void ast_rtp_codecs_payloads_unset(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, int payload); /*! + * \brief Determine the type of RTP stream media from the codecs mapped. + * \since 13.19.0 + * + * \param codecs Codecs structure to look in + * + * \return Media type or AST_MEDIA_TYPE_UNKNOWN if no codecs mapped. + */ +enum ast_media_type ast_rtp_codecs_get_stream_type(struct ast_rtp_codecs *codecs); + +/*! * \brief Retrieve payload information by payload * * \param codecs Codecs structure to look in |