VoIP Using SIP/RTP

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VoIP Using SIP/RTP

• by
• George Fu, UCCS
• CS 522 Semester Project
• Fall 2004

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Two Parts of the Project

• Understand VOIP
• Implement SIP and RTP

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Voice To/From IP
Analog
Voice
CODEC Analog to Digital
Compress
Create Voice Datagram
Add Header (RTP, UDP, IP, etc)
Digital
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Telephone-to-PC
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ISO Reference Model and VoIP Standards
 
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SIP Session Initiation Protocol

• Its a signaling protocol proposed by IETF.
• Establish sessions.
• SIP is a text-based, peer-to-peer protocol that
  runs on the Session Layer.
• SIP Address Format (resembles mailto URL format)
• siphenrys_at_mci.com
• sip 1-972-555-1234_at_mci.com userphone
• Integrated heavily w/ Internet technologies such
  as web (http), email messaging services, and
  directory services (DNS).
• Location Independent and hence opted for Mobile
  Networks.

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SIP Architecture

• Major Entities
• User Agent
• Intermediate Server
• Proxy Server
• Redirect Server
• SIP Registrar
• Gateway

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SIP Messages Methods and Responses
SIP components communicate by exchanging SIP
messages

• SIP Methods
• INVITE Initiates a call by inviting user to
  participate in session.
• ACK - Confirms that the client has received a
  final response to an INVITE request.
• BYE - Indicates termination of the call.
• CANCEL - Cancels a pending request.
• REGISTER Registers the user agent.
• OPTIONS Used to query the capabilities of a
  server.
• INFO Used to carry out-of-bound information,
  such as DTMF digits.

• SIP Responses
• 1xx - Informational Messages.
• 2xx - Successful Responses.
• 3xx - Redirection Responses.
• 4xx - Request Failure Responses.
• 5xx - Server Failure Responses.
• 6xx - Global Failures Responses.

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Example of SIP message

• INVITE sipbob_at_domain.com SIP/2.0
• Via SIP/2.0/UDP 166.34.27.44
• From sipalice_at_mci.com
• To sipbob_at_domain.com
• Call-ID a2e3a_at_mci.com
• Content-Type application/sdp
• Content-Length 885
• cIN IP4 166.34.27.44
• maudio 38060 RTP/AVP 0

• HTTP message syntax
• sdp session description protocol
• Call-ID is unique for every call.

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Overview of RTP

• Provides end-to-end delivery services for
  real-time traffic interactive audio and video.
• Payload identification, sequence numbering,
  time-stamping and delivery monitoring.
• Runs on top of UDP, and less often, TCP.
• RTP does not guarantee delivery or prevent
  out-of-order delivery.

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PC-to-PC
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Call to a known Computer

• Alices SIP invite message indicates her port
  number IP address. Indicates encoding that
  Alice prefers to receive (PCM ulaw)
• Bobs 200 OK message indicates his port number,
  IP address preferred encoding (GSM)
• SIP messages can be sent over TCP or UDP here
  sent over RTP/UDP.
• Default SIP port number is 5060.

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Future Work
Delay For high quality voice, one way latency must not be greater than 150ms. Delay greater than 50ms leads to echo and talker overlap.
Jitter Variation in inter-packet arrival time. The solution to this problem is to introduce jitter buffers.
Packet Loss Loss in excess of 5-10 causes significant degradation in voice quality.
Re-ordering Packets may arrive out of order and this leads to garbled speech.
Speech Coding PCM, PCM uLaw, ADPCM, LPC, LD-CELP, GSM
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References

• U. Black, Voice over IP, 2nd ed., Prentice Hall,
  2002
• J. Davidson and J. Peters, Voice over IP
  Fundamentals, Cisco Press, 2000
• Douskalis, IP Telephony. The Integration of
  Robust IP Services, Prentice Hall, 2000.
• H. Liu and P. Mouchtaris, Voice over IP
  Signaling H.323 and Beyond, IEEE Comm. Mag.,
  October 2000, pp. 142-148
• H. Schulzrinne and J. Rosenberg, The Session
  Initiation Protocol Internet-Centric Signaling,
  IEEE Commun. Mag., Oct. 2000, pp. 134-141.
• RFC 1889 H. Schulzrinne et al, RTP A Transport
  Protocol for Real-Time Applications
• http//www.itpapers.com/techguide/voiceip.pdf
• http//www.cs.columbia.edu/sip/