The RTP protocol, being designed for ``real-time'' transport of media, behaves like a streaming protocol should. RTP datagram packets are relatively small and there are usually tens to hundreds of packets sent per second in the process of relaying audio between two peers. Additionally, different audio Codecs provide for different encoded audio sample sizes, resulting in a variable amount of available space for embedding which is dependent upon which Codec the audio for any individual RTP packet is encoded with. Due to the small size of these packets and the common constraint among many steganographic embedding methods which limits the amount of data that is able to be embedded to a fraction of the size of the cover-medium, a very limited amount of space is actually available for the embedding of message data. As such, large message data will inevitably be required to be split across multiple cover-packets and thus must be reassembled at its destination.