[Asterisk-code-review] res rtp asterisk.c: Disable packet flood detection for video... (asterisk[13])
Jenkins2
asteriskteam at digium.com
Fri Dec 15 11:59:54 CST 2017
Jenkins2 has submitted this change and it was merged. ( https://gerrit.asterisk.org/7573 )
Change subject: res_rtp_asterisk.c: Disable packet flood detection for video streams.
......................................................................
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
---
M configs/samples/rtp.conf.sample
M include/asterisk/rtp_engine.h
M main/rtp_engine.c
M res/res_rtp_asterisk.c
4 files changed, 73 insertions(+), 14 deletions(-)
Approvals:
Joshua Colp: Looks good to me, but someone else must approve
George Joseph: Looks good to me, approved
Jenkins2: Approved for Submit
diff --git a/configs/samples/rtp.conf.sample b/configs/samples/rtp.conf.sample
index 9bc3de3..de9d590 100644
--- a/configs/samples/rtp.conf.sample
+++ b/configs/samples/rtp.conf.sample
@@ -21,9 +21,17 @@
; rtcpinterval = 5000 ; Milliseconds between rtcp reports
;(min 500, max 60000, default 5000)
;
-; Enable strict RTP protection. This will drop RTP packets that
-; do not come from the source of the RTP stream. This option is
-; enabled by default.
+; Enable strict RTP protection. This will drop RTP packets that do not come
+; from the recoginized source of the RTP stream. Strict RTP qualifies RTP
+; packet stream sources before accepting them upon initial connection and
+; when the connection is renegotiated (e.g., transfers and direct media).
+; Initial connection and renegotiation starts a learning mode to qualify
+; stream source addresses. Once Asterisk has recognized a stream it will
+; allow other streams to qualify and replace the current stream for 5
+; seconds after starting learning mode. Once learning mode completes the
+; current stream is locked in and cannot change until the next
+; renegotiation.
+; This option is enabled by default.
; strictrtp=yes
;
; Number of packets containing consecutive sequence values needed
diff --git a/include/asterisk/rtp_engine.h b/include/asterisk/rtp_engine.h
index 0b29f34..7daff67 100644
--- a/include/asterisk/rtp_engine.h
+++ b/include/asterisk/rtp_engine.h
@@ -1348,6 +1348,16 @@
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
diff --git a/main/rtp_engine.c b/main/rtp_engine.c
index e703272..b12761b 100644
--- a/main/rtp_engine.c
+++ b/main/rtp_engine.c
@@ -887,6 +887,25 @@
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->payloads); ++payload) {
+ type = AST_VECTOR_GET(&codecs->payloads, 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;
diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c
index cf793a6..230d147 100644
--- a/res/res_rtp_asterisk.c
+++ b/res/res_rtp_asterisk.c
@@ -255,6 +255,8 @@
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 @@
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 @@
* 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",
--
To view, visit https://gerrit.asterisk.org/7573
To unsubscribe, visit https://gerrit.asterisk.org/settings
Gerrit-Project: asterisk
Gerrit-Branch: 13
Gerrit-MessageType: merged
Gerrit-Change-Id: I78031491a6e75c2d4b1e9c2462dc498fe9880a70
Gerrit-Change-Number: 7573
Gerrit-PatchSet: 2
Gerrit-Owner: Richard Mudgett <rmudgett at digium.com>
Gerrit-Reviewer: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: Jenkins2
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Richard Mudgett <rmudgett at digium.com>
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