45 Media Transfer and Codecs Media Transfer H323 and SIP are only the signaling

45 media transfer and codecs media transfer h323 and

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4.5 Media Transfer and Codecs Media Transfer H.323 and SIP are only the signaling protocols. Audio data is carried by RTP [8] (Real-time Transport Protocol). The RTP protocol provides features for real-time applications, with the ability to reconstruct timing, loss detection, security, content delivery and identification of encoding schemes. The media gateways that digitize voice use the RTP protocol to deliver the voice (bearer) traffic. For each participant, a particular pair of destination IP addresses defines the session between the two endpoints, which translate into a single RTP session for each phone call in progress. Again three network setups in Skype are discussed. First, if both Skype clients are on public IP address, then media traffic flows directly between them over UDP. Second, if either caller or callee or both were behind port-restricted NAT, they sent voice traffic to another online Skype node over UDP. That node acted as a media proxy and forwarded the voice traffic from caller to callee and vice versa. At last, if both users are behind port-restricted NAT and UDP-restricted firewall, then caller and callee send and receive voice traffic over TCP from another online Skype node. The Skype protocol seems to prefer the use of UDP for voice transmission as much as possible. The SC will use UDP for voice transmission if it is behind a NAT or firewall that allows UDP packets to flow across.
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INFOTECH Seminar Advanced Communication Services (ACS), 2005 9 Codecs Voice communication is analog, while data networking is digital. The process of converting analog waveforms to digital information is done with a coder-decoder. There are many ways an analog voice signal can be transformed, all of which are governed by various standards. The process of conversion is complex and beyond the scope of this paper. Suffice to say that most of the conversions are base on pulse coded modulation (PCM) or variations. In addition to performing the analog to digital conversion, Codecs compress the data stream, and provide echo cancellation. Compression of the represented waveform can lead to bandwidth savings. Another way to save bandwidth is the use of silence suppression, which is the process of not sending voice packets between the gaps in human conversations. Global IP Sound [9] provides voice processing software to Skype. Skype automatically selects the best codec depending on the connection. All solutions are designed specifically for use on packet networks: - No inter-frame dependency - Multiple description coding - Variable bit-rate It should be known that no silence suppression is supported in Skype. Henning Schulzrinne observed that when neither caller nor callee was speaking, voice packets still flowed between them. Transmitting these silence packets has two advantages. First, it maintains the UDP bindings at NAT and second, these packets can be used to play some background noise at the peer. In the case where media traffic flowed over TCP between caller and callee, silence packets were still sent. The purpose is to avoid the drop in TCP congestion window size, which
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