Voice Over IP

1
Voice Over IP

• Mehran Dowlatshahi
• Faculty of IT
• UTS

http//www-staff.it.uts.edu.au/mehdow/
2
Overview

• Overview of VOIP Layers
• Bandwidth for Different VOIP Codecs
• VOIP Scenarios
• VOIP Standards
• Conclusion

3
Voice To IP
Codec
Analog
Sample and A/D
Compress
Create Voice Datagram (RTP,UDP)
Add IP Header
Digital
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Voice From IP
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UDP (User Datagram Protocol)

• Source port (2 Bytes)
• Identifies the higher layer process which
  originated the data.
• Destination port (2 Bytes)
• Identifies with higher layer process to which
  this data is being transmitted.
• Length (2 Bytes)
• The length in octets of the UDP data and payload
  (minimum 8).
• Checksum (2 Bytes)
• Optional field supporting error detection.

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RTP (Real-time Transport Protocol)

• Version
• Identifies the version of RTP (currently version
  2).
• Sequence number
• A unique reference number which increments by one
  for each RTP packet sent. It allows the receiver
  to reconstruct the sender's packet sequence.
• Timestamp
• The time that this packet was transmitted. This
  field allows the received to buffer and playout
  the data in a continuous stream.
• Payload Type
• Synchronization Source SSRC Source of RTP Stream
  (32 bits)

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RTP,UDP,IP Overhead

• 40 octet overhead for every packet of data
• The more Voice Samples per Packet the more
  Efficient
• For voice, samples representing 20ms are
  considered the maximum duration for the payload.

8
Voice Bandwidth

• Coding, Bandwidth, Sample IP bandwidth
• G.711 PCM1 64kbps 0.125ms 80kbps
• G.723.1 ACELP2 5.6kbps 30ms 16.27kbps 6.4kbps
  17.07kbps
• G.726 ADPCM1 32kbps 0.125ms 48kbps
• G.728 LD-CELP2 16kbps 0.625ms 32kbps
• G.729(A) CS-ACELP2 8kbps 10ms 24kbps
• 1- Adaptive Differential / Pulse Code Modulation
  (AD / PCM)
• 2- Conjugate Structure/Algebric/Low Delay/ Code
  Excited Linear Prediction (CS-A/A/LD/ CELP)

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Scenario 1 PC to PC

• Need a PC with sound card
• IP Telephony software Internet Phone,
  Netmeeting, ...
• Video optional
• Access to IP Network

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Scenario 2 PC to Phone

• Need a gateway that connects IP Network to phone
  Network

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Scenario 3 Phone to Phone

• Need Two Gateways one Close to Caller and One
  Close to Called Party

12
VOIP Standards

• H.323 (ITU-T Umbrella of Standards for Multimedia
  Communication)
• Call setup and closure, Capability Exchange,
  E.164 to Gatekeeper address translation,
  Admission Control, Conferencing, Media
  encodingdecoding.
• SIP (Session Initiation Protocol, IETF RFC2543) A
  Simple and Extensible Protocol,

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VOIP Protocol Suites
VOIP Coding
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H323
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H.323 Call Stages
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H323 Gatekeeper

• Mandatory Gatekeeper Functions Are
• Address Translation number to IP Address
• Admission Control Controls endpoint admission
  into the H.323 network. uses the followingH.225
  Registration, Admission, and Status (RAS)
  messages
• Bandwidth Control Accepts/Rejects BW Request
• Zone Management - Controlling the endpoint (GW)
  registration process.

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SIP Interactions
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SIP Simple Operation
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Conclusions

• In terms of functionality and services that can
  be supported, H.323 version 2 and SIP are very
  similar.
• Supplementary services in H.323 are more
  rigorously defined. Therefore, fewer
  interoperability issues are expected among its
  implementations.
• H.323 has better compatibility among its
  different versions and better interoperability
  with PSTN. The two protocols are comparable in
  their QoS support (similar call setup delays, no
  support for resource reservation or class of
  service (QoS) setting), but H.323 version 3 will
  allow signaling of the requested QoS.
• SIP's primary advantages are its flexibility to
  add new features and its relative ease of
  implementation and debugging.
• Finally, we note that H.323 and SIP are improving
  themselves by learning from the other side, and
  the differences between them are diminishing with
  each new version.

21
Thank You

• Questions Please

http//www-staff.it.uts.edu.au/mehdow/