Porting VoIP applications to DCCP

ABSTRACT

In the digital era, the increase of network bandwidth and the ubiquitous wireless access facilitates the creation of more and more innovative network services. Among these services, Voice over Internet Protocol (VoIP) is surely one of the most popular and successful real-time multimedia services on the Internet. For decades, User Datagram Protocol (UDP) has been adopted to transport the voice streams of VoIP applications on the network. In this paper, we presented how the Datagram Congestion Control Protocol (DCCP), recently developed by Internet Engineering Task Force (IETF), can be utilized to transport realtime audio stream through Real-time Transport Protocol (RTP). We proposed the design and architecture of a VoIP application running on DCCP, and take Linphone, an open-source Internet VoIP phone on Linux, as an example to illustrate how to apply DCCP to establish a bidirectional Session Initiation Protocol (SIP) communication.

REFERENCES

• 1: H. Schulzrinne, S. Casner, R. Frederick, V. Jacobson, "RTP: A Transport Protocol for Real-Time Applications", IETF RFC 1889, January 1996.
• 2: H. Schulzrinne , S. Casner , R. Frederick , V. Jacobson, RTP: A Transport Protocol for Real-Time Applications, RFC Editor, 2003
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• 4: E. Kohler, M. Handley, S. Floyd, "Datagram Congestion Control Protocol (DCCP)", IETF RFC 4340, March 2006.
• 5: S. Floyd, E. Kohler, "Datagram Congestion Control Protocol (DCCP) Congestion Control ID 2: TCP-like Congestion Control ", IETF RFC 4341, March 2006.
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INDEX TERMS

Primary Classification:
  C. Computer Systems Organization
  C.2 COMPUTER-COMMUNICATION NETWORKS
      C.2.2 Network Protocols

General Terms:
Experimentation, Performance

Keywords:
DCCP, RTP, SIP, VoIP