VoIP基础教程

In-Network Calling
Current and Future Features – Centrex (NGN/IMS) – Multimedia Messaging – Presence – PoC (Push to Talk over Cellular) –…
Contents
ATM Asynchronous transfer mode, a cells witched communications technology
SS7: Signaling System 7 SG: Signaling gateway
H.323: An ITU-T standard protocol suite for real-time communications over a packet network.
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VoIP Basics
VoIP Architecture
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VoIP Signaling & Media
VoIP Evoluຫໍສະໝຸດ Baiduion
ITU VoIP Model
IETF VoIP Model
The Whole VoIP Architecture
Contents
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VoIP Basics
– Jitter
– Packet loss – Bandwidth
• Bandwidth sharing between voice and data may worse the voice
Reliability
– Network stability will make it hard to guarantee the 99.999% reliability
H.225: An ITU-T call signaling protocol (part of the H.323 suite). H.235: An ITU-T security protocol (part of the H.323 suite). H.245: An ITU-T capability exchange protocol (part of the H.323 suite). SIP: Session Initial Protocol RTP: Real-time Transport Protocol RTCP: Real-time Cotrol Protocol RTSP Real-time streaming protocol SDP Session description protocol
CS-ACELP: Conjugate Structure Algebraic-Code Excited Linear Prediction AMR: Adaptive Multi-Rate
VoIP Concepts
VoIP: Transmission of voice over Internet How VoIP works
H.248 / Megaco
ITU-T H.248 / IETF RFC 3525 Centralized Call-Control Architecture Used between Call-Agents (MGC) & Gateways (MG) Media control protocol in NGN/IMS
VoIP Architecture
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VoIP Signaling & Media
VoIP Evolution
VoIP Protocol Stack
VoIP Signaling Protocols
H.323
– – – – – – – – – – – – ITU standard, ISDN-based, distributed topology 90%+ of all Service Provider VoIP networks The current interconnect for CallManager to Service Providers Useful for video applications
– – – – – Continuously sample audio Encode each sample to digital stream Send digitized stream across Internet in packets Receive and convert packets to digital stream Decode the stream back to analog for playback
– RTP Streaming: Packetizing stream and transmitting over IP network
• RTP over UDP instead of TCP: Why?
VoIP Components
Servers – Session Controller
– Media Gateway Controller
End-point devices
– H323 Phone – SIP Phone – MGCP Endpoint
Operability Challenges
Voice Quality
– Latency
• • • VoIP typically tolerates delays up to 150 ms before the quality of the call degrades. Instantaneous buffer use causes delay variation in the same voice stream. Loss of packets severely degrades the voice application.
VoIP Tutorial
Li, Xiaomeng 2010.06.10
Contents
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VoIP Basics
VoIP Architecture
3 VoIP Standard & Protocols 4
VoIP Evolution
VoIP Terminology
VoIP: Voice Over IP PSTN: Public switched telephone network IETF: Internet Engineering Task Force ITU: International Telecommunications Union
MGCP
– IETF RFC3435 – Centralized Call-Control Architecture – Used between Call-Agents (MGC) & Gateways (MG)
H.323 Overview
An ITU recommendation applicable to “Packet-based multimedia communications systems”. H.323 defines a distributed architecture for creating multimedia applications, including VoIP H.323 consists of a set of protocols working together to handle all aspects of communication, including: – Transmission of a digital audio phone call – Signaling to set up and manage phone call – Allows transmission of video and data while a phone call is in progress – Sends binary message – Incorporates protocols for security – Uses a special hardware Multipoint Control Unit for conferencing calls – Defines servers for address resolution, authentication, accounting, features, etc Older and more established protocol
– Disadvantages: • Continuous service during a power outage • Emergency calls
• Operability
VoIP Functionalities
Signaling: the process of establishing and terminating a call • Registration, Call initiation, Call teardown • Bearer control (codec negotiation)
MG: Media gateway
SBC: Session Border Controller DTMF: Dual tone multiple frequency PCM: Pulse Code Modulation
ADPCM: Adaptive Differential Pulse Code Modulation
Scalability
Security
Features Interoperability Switch over cost
Call Features
Traditional Call Features – CLASS 5 Call Features
• • • • • Caller ID Indication Call Waiting Call Forwarding 3-Way Calling Call Baring
SIP
IETF RFC3261 with many extensive RFCs Distributed Call-Control Used for more than VoIP…SIMPLE: Instant Messaging / Presence Session control protocol in NGN/IMS
VoIP Advantages & Disadvantages – Advantages
• • • • Much lower cost (multiplexing, geographic insensitive) Portability: call at any where Extensive call features Converged voice, video, data over IP network
• Managing interaction with PBX
– Application Server Gateways – Media Gateway: transcode audio between IP network and PSTN – Signaling Gateway: translates signaling operations – Residential Gateway
Media (Voice) Streaming
– Codec: encode voice into bit stream/decode the stream to voice
• • • • • G711/PCM: mu-law, A-law, 64kbps G.723: 5.3, 6.4kbps G.726/ADPCM: 16, 24, 32, and 40kbps G.729/CS-ACELP: 8kbps AMR: Variable rate from 4.75 ~ 12.2 kbps