SIPRTCP Implementation by George Fu UCCS CS 525

SIP/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 (RTP, UDP, IP, etc) Digital

ISO Reference Model and Vo. IP 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.

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

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: a 2 e 3 a@mci. com Content-Type: application/sdp Content-Length: 885 c=IN IP 4 166. 34. 27. 44 m=audio 38060 RTP/AVP 0 • HTTP message syntax • sdp = session description protocol • Call-ID is unique for every call.

PC-to-PC

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 • Open All the Sockets in the Main Thread sip_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); rtcp_receive_socket = socket (AF_INET, SOCK_DGRAM, 0);

Implementation • Separate Thread for RTP and RTCP pthread_create(&child, NULL, send_RTP_RTCP, NULL); • Use Select System Call to Make SIP/RTCP/User Commands Send and Receive all Work Simultaneously

Demo

Future Work Delay For high quality voice, one way latency must not be greater than 150 ms. Delay greater than 50 ms 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 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 u. Law, ADPCM, LPC, LDCELP, GSM

References • U. Black, Voice over IP, 2 nd 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/