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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
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@@ -3667,7 +3667,17 @@ static struct ast_frame *__ast_read(struct ast_channel *chan, int dropaudio, int
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* originated from and update the frame to include it.
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*/
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stream = default_stream = ast_channel_get_default_stream(chan, ast_format_get_type(f->subclass.format));
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f->stream_num = ast_stream_get_position(stream);
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/* In order to allow media to be passed up the underlying media type has to have a format negotiated on
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* the channel itself. In cases where this hasn't happened the channel driver is incorrectly passing up
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* a frame for a format that has not been negotiated. If this occurs just drop the frame as we have no
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* stream that it came from.
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*/
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if (!stream) {
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ast_frfree(f);
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f = &ast_null_frame;
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} else {
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f->stream_num = ast_stream_get_position(stream);
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}
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}
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}
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} else {
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@@ -3700,7 +3710,12 @@ static struct ast_frame *__ast_read(struct ast_channel *chan, int dropaudio, int
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*/
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if (f && (f->frametype == AST_FRAME_VOICE || f->frametype == AST_FRAME_VIDEO)) {
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stream = default_stream = ast_channel_get_default_stream(chan, ast_format_get_type(f->subclass.format));
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f->stream_num = ast_stream_get_position(stream);
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if (!stream) {
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ast_frfree(f);
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f = &ast_null_frame;
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} else {
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f->stream_num = ast_stream_get_position(stream);
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}
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}
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}
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else
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@@ -1216,7 +1216,7 @@ struct ast_rtp_payload_type *ast_rtp_codecs_get_payload(struct ast_rtp_codecs *c
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}
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ast_rwlock_unlock(&codecs->codecs_lock);
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if (!type) {
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if (!type && payload <= AST_RTP_PT_LAST_STATIC) {
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ast_rwlock_rdlock(&static_RTP_PT_lock);
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type = ao2_bump(static_RTP_PT[payload]);
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ast_rwlock_unlock(&static_RTP_PT_lock);
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