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 # RTP in WebRTC

 WebRTC uses the RTP protocol described in
 [RFC3550](https://datatracker.ietf.org/doc/html/rfc3550) for transporting audio
 and video. Media is encrypted using [SRTP](./srtp.md).

 ## Allocation of payload types

 RTP packets have a payload type field that describes which media codec can be
 used to handle a packet. For some (older) codecs like PCMU the payload type is
 assigned statically as described in
 [RFC3551](https://datatracker.ietf.org/doc/html/rfc3551). For others, it is
 assigned dynamically through the SDP. **Note:** there are no guarantees on the
 stability of a payload type assignment.

 For this allocation, the range from 96 to 127 is used. When this range is
 exhausted, the allocation falls back to the range from 35 to 63 as permitted by
 [section 5.1 of RFC3550][1]. Note that older versions of WebRTC failed to
 recognize payload types in the lower range. Newer codecs (such as flexfec-03 and
 AV1) will by default be allocated in that range.

 Payload types in the range 64 to 95 are not used to avoid confusion with RTCP as
 described in [RFC5761](https://datatracker.ietf.org/doc/html/rfc5761).

 ## Allocation of audio payload types

 Audio payload types are assigned from a table by the [PayloadTypePicker][2]
 class. Audio codecs generally prefer the upper dynamic range [96,127] to
 maintain compatibility with older WebRTC versions. Redundancy formats like RED
 are allocated in the lower dynamic range [35,63], starting at 63, to reduce
 collisions with video payload types.

 ## Allocation of video payload types

 Video payload types are allocated by the
 [GetPayloadTypesAndDefaultCodecs method][3]. The set of codecs depends on the
 platform, in particular for H264 codecs and their different profiles. Payload
 numbers are assigned ascending from 96 for video codecs and their
 [associated retransmission format](https://datatracker.ietf.org/doc/html/rfc4588).
 Some codecs like flexfec-03 and AV1 are assigned to the lower range [35,63] for
 reasons explained above. When the upper range [96,127] is exhausted, payload
 types are assigned to the lower range [35,63], starting at 35.

 ## Handling of payload type collisions

 Due to the requirement that payload types must be uniquely identifiable when
 using [BUNDLE](https://datatracker.ietf.org/doc/html/rfc8829) collisions between
 the assignments of the audio and video payload types may arise. These are
 resolved by the [PayloadTypeSuggester][4] interface which will reassign payload
 type numbers descending from 127.

 # Bitrate probing

 Bandwidth estimation sometimes requires sending RTP packets to ramp up the
 estimate above a certain treshold. WebRTC uses two techniques for that:

 - Packets that only consist of RTP padding
 - RTX packets

 At the receiving end, both types of probing packets do not interfere with media
 processing. After being taken into account for bandwidth estimation, probing
 packets only consisting of padding can be dropped and RTX packets act as
 redundancy.

 Using RTX for probing is generally preferred as padding probes are limited to
 256 bytes of RTP payload which results in a larger number of packets being used
 for probing which is a disadvantage from a congestion control point of view.

 ## Padding probes

 Padding probes consist of RTP packets with header extensions (either abs-send
 time or the transport-wide-cc sequence number) and set the RTP "P" bit. The last
 byte of the RTP payload which specifies the amount of padding is set to 255 and
 preceeded by 255 bytes of all zeroes. See [section 5.1 of RFC3550][1]. Padding
 probes use the RTX RTP stream (i.e. payload type, sequence number and timestamp)
 when RTX is negotiated or share the same RTP stream as the media packets
 otherwise.

 Padding probes are used either when

 - RTX is not negotiated (such as for audio, less commonly for video) or
 - no suitable original packet is found for RTX probing.

 Padding probes should not be interleaved with packets of a video frame.

 ## RTX probes

 RTX probes are resends of previous packets that use RTX retransmissions
 specified in [RFC4588](https://www.rfc-editor.org/rfc/rfc4588) as payload format
 when negotiated with the peer. These packets will have a different abs-send-time
 or transport-wide-cc sequence number and use the RTX RTP stream (i.e. RTX
 payload type, sequence number and timestamp) associated with the media RTP
 stream.

 RTX probing uses recently sent RTP packets that have not yet been acknowledged
 by the remote side. Sending these packets again has a small chance of being
 useful when the original packet is lost and will not affect RTP processing at
 the receiver otherwise.

 [1]: https://datatracker.ietf.org/doc/html/rfc3550#section-5.1
 [2]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/call/payload_type_picker.h;l=35;drc=031d798a20cc5d9c867598a38632accb20c6effe
 [3]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/media/engine/webrtc_video_engine.cc;l=206;drc=031d798a20cc5d9c867598a38632accb20c6effe
 [4]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/call/payload_type.h;l=26;drc=031d798a20cc5d9c867598a38632accb20c6effe
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	# RTP in WebRTC

	WebRTC uses the RTP protocol described in
	[RFC3550](https://datatracker.ietf.org/doc/html/rfc3550) for transporting audio
	and video. Media is encrypted using [SRTP](./srtp.md).

	## Allocation of payload types

	RTP packets have a payload type field that describes which media codec can be
	used to handle a packet. For some (older) codecs like PCMU the payload type is
	assigned statically as described in
	[RFC3551](https://datatracker.ietf.org/doc/html/rfc3551). For others, it is
	assigned dynamically through the SDP. Note: there are no guarantees on the
	stability of a payload type assignment.

	For this allocation, the range from 96 to 127 is used. When this range is
	exhausted, the allocation falls back to the range from 35 to 63 as permitted by
	[section 5.1 of RFC3550][1]. Note that older versions of WebRTC failed to
	recognize payload types in the lower range. Newer codecs (such as flexfec-03 and
	AV1) will by default be allocated in that range.

	Payload types in the range 64 to 95 are not used to avoid confusion with RTCP as
	described in [RFC5761](https://datatracker.ietf.org/doc/html/rfc5761).

	## Allocation of audio payload types

	Audio payload types are assigned from a table by the [PayloadTypePicker][2]
	class. Audio codecs generally prefer the upper dynamic range [96,127] to
	maintain compatibility with older WebRTC versions. Redundancy formats like RED
	are allocated in the lower dynamic range [35,63], starting at 63, to reduce
	collisions with video payload types.

	## Allocation of video payload types

	Video payload types are allocated by the
	[GetPayloadTypesAndDefaultCodecs method][3]. The set of codecs depends on the
	platform, in particular for H264 codecs and their different profiles. Payload
	numbers are assigned ascending from 96 for video codecs and their
	[associated retransmission format](https://datatracker.ietf.org/doc/html/rfc4588).
	Some codecs like flexfec-03 and AV1 are assigned to the lower range [35,63] for
	reasons explained above. When the upper range [96,127] is exhausted, payload
	types are assigned to the lower range [35,63], starting at 35.

	## Handling of payload type collisions

	Due to the requirement that payload types must be uniquely identifiable when
	using [BUNDLE](https://datatracker.ietf.org/doc/html/rfc8829) collisions between
	the assignments of the audio and video payload types may arise. These are
	resolved by the [PayloadTypeSuggester][4] interface which will reassign payload
	type numbers descending from 127.

	# Bitrate probing

	Bandwidth estimation sometimes requires sending RTP packets to ramp up the
	estimate above a certain treshold. WebRTC uses two techniques for that:

	- Packets that only consist of RTP padding
	- RTX packets

	At the receiving end, both types of probing packets do not interfere with media
	processing. After being taken into account for bandwidth estimation, probing
	packets only consisting of padding can be dropped and RTX packets act as
	redundancy.

	Using RTX for probing is generally preferred as padding probes are limited to
	256 bytes of RTP payload which results in a larger number of packets being used
	for probing which is a disadvantage from a congestion control point of view.

	## Padding probes

	Padding probes consist of RTP packets with header extensions (either abs-send
	time or the transport-wide-cc sequence number) and set the RTP "P" bit. The last
	byte of the RTP payload which specifies the amount of padding is set to 255 and
	preceeded by 255 bytes of all zeroes. See [section 5.1 of RFC3550][1]. Padding
	probes use the RTX RTP stream (i.e. payload type, sequence number and timestamp)
	when RTX is negotiated or share the same RTP stream as the media packets
	otherwise.

	Padding probes are used either when

	- RTX is not negotiated (such as for audio, less commonly for video) or
	- no suitable original packet is found for RTX probing.

	Padding probes should not be interleaved with packets of a video frame.

	## RTX probes

	RTX probes are resends of previous packets that use RTX retransmissions
	specified in [RFC4588](https://www.rfc-editor.org/rfc/rfc4588) as payload format
	when negotiated with the peer. These packets will have a different abs-send-time
	or transport-wide-cc sequence number and use the RTX RTP stream (i.e. RTX
	payload type, sequence number and timestamp) associated with the media RTP
	stream.

	RTX probing uses recently sent RTP packets that have not yet been acknowledged
	by the remote side. Sending these packets again has a small chance of being
	useful when the original packet is lost and will not affect RTP processing at
	the receiver otherwise.

	[1]: https://datatracker.ietf.org/doc/html/rfc3550#section-5.1
	[2]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/call/payload_type_picker.h;l=35;drc=031d798a20cc5d9c867598a38632accb20c6effe
	[3]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/media/engine/webrtc_video_engine.cc;l=206;drc=031d798a20cc5d9c867598a38632accb20c6effe
	[4]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/call/payload_type.h;l=26;drc=031d798a20cc5d9c867598a38632accb20c6effe