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 /res | |
| 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 'res')
| -rw-r--r-- | res/res_rtp_asterisk.c | 44 |
1 files changed, 33 insertions, 11 deletions
diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c index cf793a663..230d147be 100644 --- a/res/res_rtp_asterisk.c +++ b/res/res_rtp_asterisk.c @@ -255,6 +255,8 @@ struct rtp_learning_info { struct timeval received; /*!< The time of the first received packet */ int max_seq; /*!< The highest sequence number received */ int packets; /*!< The number of remaining packets before the source is accepted */ + /*! Type of media stream carried by the RTP instance */ + enum ast_media_type stream_type; }; #ifdef HAVE_OPENSSL_SRTP @@ -2812,18 +2814,29 @@ static int rtp_learning_rtp_seq_update(struct rtp_learning_info *info, uint16_t info->received = ast_tvnow(); } - /* - * Protect against packet floods by checking that we - * received the packet sequence in at least the minimum - * allowed time. - */ - if (ast_tvzero(info->received)) { - info->received = ast_tvnow(); - } else if (!info->packets && (ast_tvdiff_ms(ast_tvnow(), info->received) < learning_min_duration )) { - /* Packet flood; reset */ - info->packets = learning_min_sequential - 1; - info->received = ast_tvnow(); + switch (info->stream_type) { + case AST_MEDIA_TYPE_UNKNOWN: + case AST_MEDIA_TYPE_AUDIO: + /* + * Protect against packet floods by checking that we + * received the packet sequence in at least the minimum + * allowed time. + */ + if (ast_tvzero(info->received)) { + info->received = ast_tvnow(); + } else if (!info->packets + && ast_tvdiff_ms(ast_tvnow(), info->received) < learning_min_duration) { + /* Packet flood; reset */ + info->packets = learning_min_sequential - 1; + info->received = ast_tvnow(); + } + break; + case AST_MEDIA_TYPE_VIDEO: + case AST_MEDIA_TYPE_IMAGE: + case AST_MEDIA_TYPE_TEXT: + break; } + info->max_seq = seq; return info->packets; @@ -5430,6 +5443,15 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc * source and we should switch to it. */ if (!ast_sockaddr_cmp(&rtp->rtp_source_learn.proposed_address, &addr)) { + if (rtp->rtp_source_learn.stream_type == AST_MEDIA_TYPE_UNKNOWN) { + struct ast_rtp_codecs *codecs; + + codecs = ast_rtp_instance_get_codecs(instance); + rtp->rtp_source_learn.stream_type = + ast_rtp_codecs_get_stream_type(codecs); + ast_verb(4, "%p -- Strict RTP qualifying stream type: %s\n", + rtp, ast_codec_media_type2str(rtp->rtp_source_learn.stream_type)); + } if (!rtp_learning_rtp_seq_update(&rtp->rtp_source_learn, seqno)) { /* Accept the new RTP stream */ ast_verb(4, "%p -- Strict RTP switching source address to %s\n", |