Mid-session Audio Codec Change

Most VoIP signaling protocols provide methods for VoIP endpoints to change the audio encoding method on the fly. Due to this functionality an RTP session may begin using one Codec and then switch to a completely different Codec mid-session. This functionality may be used for a variety of reasons including QoS metrics not being met, inclusion of a new endpoint in the call that does not support the original Codec, or any number of other reasons. Due to this dynamic nature, any steganographic system attempting to embed data into an RTP stream's packets must be able to dynamically adjust its message embedding algorithm to accommodate different Codecs' various sample sizes and layout within the RTP packet payload.