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. 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. 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.
Audio payload types are assigned from a table by the PayloadTypeMapper class. New audio codecs should be allocated in the lower dynamic range [35,63], starting at 63, to reduce collisions with payload types
Video payload types are allocated by the GetPayloadTypesAndDefaultCodecs method. 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. 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.
Due to the requirement that payload types must be uniquely identifiable when using BUNDLE collisions between the assignments of the audio and video payload types may arise. These are resolved by the UsedPayloadTypes class which will reassign payload type numbers descending from 127.
Bandwidth estimation sometimes requires sending RTP packets to ramp up the estimate above a certain treshold. WebRTC uses two techniques for that:
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 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. 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
Padding probes should not be interleaved with packets of a video frame.
RTX probes are resends of previous packets that use RTX retransmissions specified in 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.