VoIP Using SIP/RTP

1. VoIP Using SIP/RTP

2. Two Parts of the Project
• Understand VOIP
• Implement SIP and RTP

3. Voice To/From IP
Analog
Digital
Voice
CODEC: Analog to Digital
Compress
Create Voice Datagram
Add Header
(RTP, UDP, IP, etc)
Network

4. Telephone-to-PC

5. ISO Reference Model and VoIP Standards
ISO Protocol layer Protocols and standards
Presentation Codecs / Applications
Session H.323 / SIP / MGCP
Transport RTP / TCP / UDP
Network IP
Link FR, ATM, Ethernet, PPP, etc.

6. SIP: Session Initiation Protocol
• It’s 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)
– sip:henrys@mci.com
– sip: +1-972-555-1234@mci.com; user=phone
• Integrated heavily w/ Internet technologies such as web (http), email &
messaging services, and directory services (DNS).
• Location Independent and hence opted for Mobile Networks.

7. SIPArchitecture
• Major Entities
– User Agent
– Intermediate Server
• Proxy Server
• Redirect Server
– SIP Registrar
– Gateway

8. SIP Messages – Methods and
Responses
•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.
SIP components communicate by exchanging SIP messages:

9. Example of SIP message
INVITE sip:bob@domain.com SIP/2.0
Via: SIP/2.0/UDP 166.34.27.44
From: sip:alice@mci.com
To: sip:bob@domain.com
Call-ID: a2e3a@mci.com
Content-Type: application/sdp
Content-Length: 885
c=IN IP4 166.34.27.44
m=audio 38060 RTP/AVP 0
•HTTP message
syntax
•sdp = session
description
protocol
•Call-ID is unique
for every call.

10. 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.

11. PC-to-PC

12. Call to a known Computer
• Alice’s SIP invite message
indicates her port number &
IP address. Indicates
encoding that Alice prefers to
receive (PCM ulaw)
• Bob’s 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.
time time
Bob's
terminal rings
Alice
167.180.112.24
Bob
193.64.210.89
port 5060
port 38060
m Law audio
GSM
port 48753
INVITE bob@193.64.210.89
c=IN IP4 167.180.112.24
m=audio 38060 RTP/AVP 0
port 5060
200 OK
c=IN IP4 193.64.210.89
m=audio 48753 RTP/AVP 3
ACK
port 5060

14. 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

15. 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/