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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. 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

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  1. SIP/RTP/RTCP Implementation by George Fu, UCCS CS 525 Semester Project Fall 2006

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

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

  4. 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

  5. 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:

  6. 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.

  7. PC-to-PC

  8. 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.

  9. 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);

  10. 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

  11. Demo

  12. Future Work

  13. 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/

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