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SIP/RTP/RTCP Implementation. by George Fu, UCCS CS 525 Semester Project Fall 2006. Two Parts of the Project. Understand SIP, RTP and RTCP Implement SIP, RTP and RTCP. Voice To/From IP. Analog. Voice. CODEC: Analog to Digital. Compress. Create Voice Datagram. Add Header

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sip rtp rtcp implementation

SIP/RTP/RTCP Implementation

by

George Fu, UCCS

CS 525 Semester Project

Fall 2006

two parts of the project
Two Parts of the Project
  • Understand SIP, RTP and RTCP
  • Implement SIP, RTP and RTCP
voice to from ip
Voice To/From IP

Analog

Voice

CODEC: Analog to Digital

Compress

Create Voice Datagram

Add Header

(RTP, UDP, IP, etc)

Digital

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.

ISO Reference Model and VoIP Standards
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 Messages – Methods and Responses

SIP components communicate by exchanging SIP messages:

example of sip message
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.
call to a known computer
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.
implementation
Implementation
  • Open All the Sockets in the Main Thread

sip_send_socket = socket (AF_INET, SOCK_DGRAM, 0);

rtp_send_socket = socket (AF_INET, SOCK_DGRAM, 0);

rtcp_send_socket = socket (AF_INET, SOCK_DGRAM, 0);

sip_receive_socket = socket (AF_INET, SOCK_DGRAM, 0);

rtp_receive_socket = socket (AF_INET, SOCK_DGRAM, 0);

rtcp_receive_socket = socket (AF_INET, SOCK_DGRAM, 0);

implementation10
Implementation
  • Separate Thread for RTP and RTCP

pthread_create(&child, NULL, send_RTP_RTCP, NULL);

  • Use Select System Call to Make SIP/RTP/RTCP/User Commands Send and Receive all Work Simultaneously
references
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/