Voice in Packets: RTP, RTCP, Header Compression, Playout Algorithms, Terminal Requirements and Implementations
نویسنده
چکیده
RTP/RTCP protocol suite provides the means for sending packetized voice by introducing time stamps and sequence numbers in packet headers. Playout buffering is needed to re-synchronize the received voice stream. In this paper, a new adaptive playout delay adjustment algorithm is introduced. A major problem, especially on low-bandwidth links, with Voice over IP (VoIP) packets is that they include a lot of overhead. The solution is header compression, which is done on link-by-link basis. All terminals that support real time interactive voice should have enough processing power. The computational requirements of voice codecs usually increase with the voice compression ratio.
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