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

diff --git a/main/channel.c b/main/channel.c
index fc89d67cb27cc420fbf4248fb32d7efdee6f0112..c71d19b81f92fd773cd6b2314257617b8d80282c 100644 (file)
--- 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 cb604646f3e94e2e14d8f224a74badc840b3af3b..5242e811df331ff7d17065475f5183ee3f59b35b 100644 (file)
--- a/main/rtp_engine.c
+++ b/main/rtp_engine.c
@@ -1210,7 +1210,7 @@ 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);