WebRTC uses the RTP protocol described in RFC3550 for transporting audio and video. Media is encrypted using SRTP.
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.