OPNET Simulation of SIP Based IP Telephony over MPLS Network

The next generation communication system will provide high quality multimedia service in a more flexible and intelligent manner. In this paper, we propose a new SIP over MPLS network architecture to achieve this goal. To integrate SIP protocol with the traffic engineering function of MPLS network seamlessly and facilitate SIP call setup, the SIP-MPLS traffic aggregation server (TA server) is highlighted in this new architecture. Furthermore, the call admission and bandwidth renegotiation algorithms running on the TA server are investigated carefully. We rely on OPNET simulation to assess the performance of our network architecture and the capability of different call admission and bandwidth re-negotiation algorithms. The simulation results show that the algorithm of TBCA-BP is a suitable candidate for accomplishing the call admission and bandwidth re-negotiation task due to its satisfying performance, and our network architecture is able to shorten the SIP call setup delay greatly by using the TA server. Introduction IP telephony is becoming one of the most promising research areas today, because both Internet Sevice Providers (ISP) and traditional telephony companies can benefit from this technology. There are two highly concerned issues on using IP network to carry audio/video traffic. The first one is the signaling protocol to control a call; the second one is how IP network provides real-time audio/video applications with QoS guaranteed service. As for the first issue, Session Initiation Protocol (SIP) is proposed by Internet Engineering Task Force (IETF) as a standard for IP telephony [1,2,3]. In the panorama of protocols for IP telephony, SIP is not the only possible choice. For instance, ITU-T H.323 is an alternative candidate. However, SIP has recently gained the increasing interest from universities, standardization organizations and companies. One of the main drivers for this fact is the decision of 3GPP (Third-Generation Partnership Project) that, in the year 2000, SIP was selected as the call control protocol for 3G IP-based mobile networks. As for the second issue, Multi-protocol Label Switching (MPLS) is suggested as an apt switching technology for future core networks [4,5]. In MPLS network, the packets are labeled at the edge of the network and then routed through the network based on simple labels. This enables explicit routing and differentiated treatment of packets while keeping the core routers simple. Despite the fact that MPLS development started with goal to expedite packet forwarding, the main benefit from MPLS in current network environment is from its traffic engineering capabilities including QoS guarantee, efficient usage of network resources, resilience, and economy [6,7]. In fact, the traffic engineering capabilities make MPLS network a good choice for carrying SIP signaled audio/video traffics. Although the architecture of SIP over MPLS has been discussed in [8,9], according to our study, directly using SIP protocol over traffic engineering (TE) enabled MPLS network as previous literature suggested can cause a long delay when the SIP call connection is set up. We reach this conclusion by theoretical analysis and OPNET simulation. In the TE enabled MPLS network, when a new SIP call request comes, usually CR-LDP or RSVP-TE will be employed to set up a corresponding LSP dynamically. Our study shows that the dynamic LSP setup by CR-LDP or RSVP-TE can contribute a lot to the whole SIP call setup delay in TE enabled MPLS network. For the above reason, we propose a new network architecture of SIP over MPLS to solve this problem. As a crucial part of our architecture, the SIP-MPLS traffic aggregation server (TA server) is highlighted in this paper. The TA servers exchange traffic engineering signaling with MPLS network on behalf of a cluster of SIP clients. As a result, the SIP call setup delay in MPLS network is decreased greatly. The rest of the paper is organized as follows. Firstly, we discuss the SIP call setup delay problem existed in SIP over MPLS technology in Section 2. Then, to solve this problem, we propose a new SIP over MPLS network architecture in Section 3. Furthermore, the call admission and bandwidth re-negotiation algorithms running on TA servers are suggested in Section 4, and the OPNET simulation results of these algorithms are presented in Section 5. In the end, Section 6 summarizes our results. The SIP Call Setup Delay Problem in SIP over MPLS Technology The SIP protocol has been defined by the IETF as a signaling protocol to initiate voice, video, and multimedia sessions, and it is an important candidate for call setup signaling in IP telephony. SIP is a very flexible protocol, because it can easily work together with QoS reservation and/or admission control mechanisms.