A VoIP converged environment can be achieved through the supports that are proposed in this thesis. We have presented a simple and flexible framework for the interworking functions of VoIP protocols based on IN half-call BCSM. In addition, we have implemented the basic gateway components, O_BCSMs and T_BCSMs, for SIP, H.323, and MGCP. Using these components, gateways for SIP/H.323, SIP/MGCP, and MGCP/H.323 can be constructed. We have demonstrated a novel solution to the NAT traversal problem using SIP as an example.
Our method works no matter whether a public SIP proxy is present or not. Most important of all, it requires no changes to the existing VoIP software or devices, such as SIP user agents, SIP proxies, and NATs.
Also, we have demonstrated how those fast handoff techniques in different layers can be integrated to reduce the overall handoff delay. Our result shows that the handoff delay meets the delay requirements of VoIP applications if the MN can perform both address configuration and registration beforehand. We have analyzed three per-user location database checkpointing algorithms using numeric analysis and computer simulation. It is convinced that there may not need a sophisticated checkpointing algorithm if the checkpoint cost and paging cost are not fairly equal.
7.1 Integrated call agent
Our approach using half-call model simplifies the effort in interworking with a call signaling protocol, such as ISUP and Q.931, in the network. By using the same interaction events of the half-call model, the BCSMs of a call signaling protocol can interwork with the existing BCSMs. In addition, an ICA containing all the BCSMs is able to translate messages between call signaling protocols. Under this half-call control framework, a converged VoIP network can be managed by a group of coordinating ICAs such that two user devices managed by different ICAs can communicate with each other. The call routing function that
determines the location and protocol of the called party has not been fully investigated in this paper. As a mobile user may change his IP address and VoIP devices constantly, this problem becomes even more complicated. We need registration and/or paging schemes to track mobile users in the converged telecommunication network. Recently, P2P (peer-to-peer) VoIP communications, such as Skype, have become very popular. The interworking function for a P2P VoIP system and a client-server one (such as SIP) is an important issue that needs to be investigated.
7.2 VoIP's NAT traversal
Although our NAT traversal solution introduces an internal SIP proxy and a relay agent, their main job is to relay packets and they are easy to implement. Though the internal SIP proxy is application dependent, it inspects only a small subset of the protocol messages, such as REGISTER, INVITE, OK, and ACK in SIP. This is a practicable solution that can adapt to most VoIP applications and protocols. We also simplify the NAT type test and reduce the unnecessary hub count of the RTP stream since our IPX participates only in call setup signaling but not in RTP exchanging. The keep-alive traffic are also minimized and confined between an IPX and an RA. Although the signaling messages was relayed by an IPX and an RA, they are lightweight and do not cause vast burden.
7.3 Handoff
The handover latency of the conventional SIP-based real-time communication application such as VoIP is about 6 seconds before applying our mechanisms. By employing our designs, the latency can be reduced to about 40ms for link-layer handovers and 120ms for network and application-layer handovers, respectively. The implementation results demonstrate that the presented mechanisms consider both software designs and cross-layer optimizations of handovers, and achieve low latency handovers for wireless real-time communications.
We have reviewed several cross-layer techniques that aim to cut down handoff delays in IEEE 802.11/Mobile IP environments. Among them, a post-handoff layer-2 trigger can
successfully eliminate the link-switch detection delay. A pre-handoff layer-2 trigger, on the other hand, can be used as a signal to execute layer-3 handoff related activities prior to the associated layer-2 handoff. An MN may utilize cross-layer topology together with its location and moving direction information to determine the next serving AP and the associated FA or DHCP server to speedup both layer-2 and layer-3 handoffs.
7.4 HLR
Checkpointing can be used to enhance the reliability of the location database of PCS networks, Since each user exhibits a unique calling and moving behavior, per-user checkpointing schemes can better serve both the users and operators. In this paper, we have analyzed three per-user location database checkpointing algorithms using numeric analysis and computer simulation. The costs that we considered include the checkpointing cost and the paging cost. Our results indicate that when inter-registration times are exponentially distributed, a user location record should either be always checkpointed at registration, or be never checkpointed at all, depending on the weighting ratio between the checkpointing cost and the paging cost. If the checkpointing cost is of more concern, the user record should never be checkpointed; otherwise, the user record should always be checkpointed (duplicated) at registration. We have also studied the effects of the variance of registration interval using computer simulation. When the checkpointing cost and the paging cost almost balance, and the variance of registration interval is large, a simple checkpointing algorithm using a fixed checkpointing timer is preferred and the optimal choice of the checkpointing timer can be determined by computer simulation. In this paper, we did not investigate the effects of incoming call arrivals on the optimal choice of the checkpointing frequency directly; we assumed that the expected paging cost is known.