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This detailed overview explores the fundamentals of Voice over IP (VoIP) networking, focusing on delay challenges and the key protocols involved. It covers acceptable one-way delays for voice communication, emphasizing how delays over 400 ms are unacceptable for sensitive applications. The document discusses major VoIP protocols such as H.323, SIP, and RTP, as well as bandwidth consumption based on different coding algorithms. Additionally, it addresses key Quality of Service (QoS) solutions including packet classification, bandwidth management, and latency reduction techniques.
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IP Networking Overview • IP assumes delay and packet ordering problems
Voice Delay with IP One-Way, Mouth-to-Ear Delay Description 1-150 ms Suitable for most delay sensitive applications like voice. 150-400 ms Still acceptable, but starts to influence voice quality. Communication can be annoying if no actions are taken. 400 ms and greater Unacceptable for most delay sensitive applications. • Delay budget deals with one-way delay, not a round-trip!
Major VoIP Protocols VoIP Protocol Description H.323 • ITU standard protocol for interactive conferencing. • Evolved from H.320 ISDN standard. • Flexible, complex. Media Gateway Control Protocol (MGCP) • Emerging standard for Public Switched Telephone Network (PSTN) gateway control, thin device control. Session Initiation Protocol (SIP) • IETF protocol for interactive and non-interactive conferencing. • Simpler, but less mature, than H.323. Real-Time Transport Protocol (RTP) • IETF standard media streaming protocol. Real-Time Transport Control Protocol (RTCP) • Protocol that provides out-of-band control information for an RTP flow. Low Latency Queuing (LLQ) • Priority queuing technique that uses priority queuing-class-based weighted fair queuing (PQ-CBWFQ).
Signaling from PBX to the Router The PBX seizes a trunk line to the router, and forwards dial digits.
Signaling Between Routers The Dial Plan Mapper reads the dial digits, and finds the address of the remote IP peer. The H.323 agent initiates a Q.931 call to the remote peer.
Signaling from Router to PBX The remote H.323 agent seizes a PBX trunk, returns a Q.931 acknowledgment to the origin, and forwards dial digits to the PBX
VoIP Bandwidth Consumption Coded Frame Size (bytes) Voice Frames in VoIP Packet Layer 2 Header Size (bytes) IP+UDP+RTP Header Size (bytes) Voice Bandwidth (Kbps) Layer 2 Technology Used Total Bandwidth Required Coding Algorithm G.711 64 160 2 40 Ethernet II 26 90.4 G.711 64 160 2 40 MLPPP 7 82.8 G.711 64 160 2 2 (cRTP) MLPPP 7 67.6 Ethernet II (Cisco Default) G.729 8 20 2 40 26 34.4 G.729 8 20 2 40 MLPPP 7 26.8 G.729 8 20 2 2 (cRTP) MLPPP 7 11.6
VoIP QoS Building Blocks Devices Affected IP Network Issue Solutions Backbone speed and scale • Backbone routers • High-performance routers Packet classification • Edge routers • IP Precedence Bandwidth management and admission control • Edge routers • Voice compression • Real-Time Transport Protocol (RTP) header compression • Gatekeepers Congestion management • Edge routers • Backbone routers • Weighted Random Early Detection (WRED) Queue management • Edge routers • Backbone routers • Low Latency Queuing (LLQ) • Other queuing techniques
RTP Header Compression RTP Headers are almost always 2 Bytes long when compressed. RTP header compression saves bandwidth by compressing packet headers across WAN links.
Precedence Definition 7 Network control 6 Internetwork control 5 Critical 4 Flash-override 3 Flash 2 Immediate 1 Priority 0 Routine Configuring IP Precedence