From e70c4ec84d3553f0a979fdd5a7ca6fc1da15eabd Mon Sep 17 00:00:00 2001
From: Joshua Colp <jcolp@digium.com>
Date: Sat, 13 Jan 2018 14:04:33 +0000
Subject: [PATCH] AST-2018-001: rtp / channel: Don't allow an unnegotiated
format to be passed up.
When an RTP packet is received by an RTP engine it has to map the
payload into the Asterisk format. The code was incorrectly checking
our own static list for ALL payloads if it couldn't find a negotiated one.
This included dynamic payloads. If the payload mapped to a format
of a different type (for example receiving a video packet on an audio
RTP instance) then the core stream code could cause a crash if a legacy
channel driver was in use as no stream would be present.
To provide further protection the core stream code will no longer assume
that a video or audio frame will always have a stream for legacy channel
drivers. If no stream is present the frame is dropped.
ASTERISK-27488
Change-Id: I022556f524ad8379ee73f14037040af17ea3316a
---
main/channel.c | 19 +++++++++++++++++--
main/rtp_engine.c | 2 +-
2 files changed, 18 insertions(+), 3 deletions(-)
diff --git a/main/channel.c b/main/channel.c
index fc89d67cb2..c71d19b81f 100644
--- a/main/channel.c
+++ b/main/channel.c
@@ -3667,7 +3667,17 @@ static struct ast_frame *__ast_read(struct ast_channel *chan, int dropaudio, int
* originated from and update the frame to include it.
*/
stream = default_stream = ast_channel_get_default_stream(chan, ast_format_get_type(f->subclass.format));
- f->stream_num = ast_stream_get_position(stream);
+ /* In order to allow media to be passed up the underlying media type has to have a format negotiated on
+ * the channel itself. In cases where this hasn't happened the channel driver is incorrectly passing up
+ * a frame for a format that has not been negotiated. If this occurs just drop the frame as we have no
+ * stream that it came from.
+ */
+ if (!stream) {
+ ast_frfree(f);
+ f = &ast_null_frame;
+ } else {
+ f->stream_num = ast_stream_get_position(stream);
+ }
}
}
} else {
@@ -3700,7 +3710,12 @@ static struct ast_frame *__ast_read(struct ast_channel *chan, int dropaudio, int
*/
if (f && (f->frametype == AST_FRAME_VOICE || f->frametype == AST_FRAME_VIDEO)) {
stream = default_stream = ast_channel_get_default_stream(chan, ast_format_get_type(f->subclass.format));
- f->stream_num = ast_stream_get_position(stream);
+ if (!stream) {
+ ast_frfree(f);
+ f = &ast_null_frame;
+ } else {
+ f->stream_num = ast_stream_get_position(stream);
+ }
}
}
else
diff --git a/main/rtp_engine.c b/main/rtp_engine.c
index f108a703ba..627605a1a8 100644
--- a/main/rtp_engine.c
+++ b/main/rtp_engine.c
@@ -1216,7 +1216,7 @@ struct ast_rtp_payload_type *ast_rtp_codecs_get_payload(struct ast_rtp_codecs *c
}
ast_rwlock_unlock(&codecs->codecs_lock);
- if (!type) {
+ if (!type && payload <= AST_RTP_PT_LAST_STATIC) {
ast_rwlock_rdlock(&static_RTP_PT_lock);
type = ao2_bump(static_RTP_PT[payload]);
ast_rwlock_unlock(&static_RTP_PT_lock);
--
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