Because of the best-effort service in Internet, direct routing path of Internet may not always meet the VoIP quality requirements. Thus, many researches proposed Peer-to-Peer VoIP systems such as SIP+P2P system, which...Because of the best-effort service in Internet, direct routing path of Internet may not always meet the VoIP quality requirements. Thus, many researches proposed Peer-to-Peer VoIP systems such as SIP+P2P system, which uses relay node to relay RTP stream from the source node to the destination node and uses application-layer routing scheme to lookup the best relay nodes. The key of those systems is how to lookup the appropriate relay nodes, which we call relay lookup problem. This paper presents a novel peer relay lookup scheme based on SIP+P2P system. The main ideas are to organize the P2P network using a Cluster overlay and to use topology-aware to optimize relay selection. We introduce the mechanism in detail, and then evaluate this mechanism in NS2 network simulation environment. The results show that our scheme is scalable and can get high relay hit ratio, which confirm the feasibility of a real system. We also make comparison with traditional schemes and the results show that our scheme has good path quality.展开更多
针对集中式基于网际互连协议的语音传输(Voice over Internet Protocol,VoIP)语音通信交换系统(Voice Communication Switching System,VCCS)的互操作性需求,提出一种基于VoIP技术的中继通信解决方案。在集中式VoIP通信架构的基础上构...针对集中式基于网际互连协议的语音传输(Voice over Internet Protocol,VoIP)语音通信交换系统(Voice Communication Switching System,VCCS)的互操作性需求,提出一种基于VoIP技术的中继通信解决方案。在集中式VoIP通信架构的基础上构建中继通信模型,结合局向管理、冠号管理、路由管理以及会议管理等模块的协同运行给出VoIP中继通信的业务处理流程,并详细阐述中继会议的核心模块。最后,通过两种异型语音通信交换系统的中继通信对比测试,验证了文章提出的VoIP中继通信方法的可靠性和互操作性。展开更多
VoIP (Voice over IP) is a rapidly growing area with great market potential. To promote it for both commercial and research purposes, a prototype VoIP system based on state-of-the-art Motorola communication techniques ...VoIP (Voice over IP) is a rapidly growing area with great market potential. To promote it for both commercial and research purposes, a prototype VoIP system based on state-of-the-art Motorola communication techniques has been developed. It is a gateway system integrating a PBX and a VoIP module. All components that H.323 defines to support VoIP are implemented in the VoIP module, though in a simplified manner. As an embedded system, the system features real timeness and task distributiveness. A number of additional techniques are used to improve the performance, including noise suppression, zero copy, and buffer structure optimization. When refined in interoperability, the system will also readily serve as a product.展开更多
In this paper, the capability of IEEE 802.11b Distributed Coordination Function (DCF) mode supporting of Constant Bit Rate (CBR) and Variable Bit Rate (VBR) Voice over IP (VoIP) traffic is investigated. Then, ...In this paper, the capability of IEEE 802.11b Distributed Coordination Function (DCF) mode supporting of Constant Bit Rate (CBR) and Variable Bit Rate (VBR) Voice over IP (VoIP) traffic is investigated. Then, the capacity of 802. lib Wireless Local Area Network (WLAN) system carrying zoice calls in a wide range of scenarios, including varying delay and packet loss rate constraints is analyzed and evaluated. Both G. 711 and G. 729 voice encoding schemes and a range of voice inter-arrival time are considered. The analyses and simulation results show that capacity is highly sensitive to the delay budget allocated to the sum of pucketization and wireless netzoork delays. For a given packet loss rate constrained, G. 729 is shown to have a capacity greater than that when G. 711 is used. The simulation results show that by supporting VBR under DCF mode the network has the approximately twice much capacity as supporting CBR has, regardless of the encoding schemes and the inter-arrival time.展开更多
随着互联网技术的进步,语音通信也经过了技术变革。基于IP的语音传输(Voice over Internet Protocol,VoIP)技术在近年来成为社会关注的热点话题,逐渐广泛应用于日常生产和生活。VoIP语音通信技术不仅信息传输质量较好,还满足了目前的语...随着互联网技术的进步,语音通信也经过了技术变革。基于IP的语音传输(Voice over Internet Protocol,VoIP)技术在近年来成为社会关注的热点话题,逐渐广泛应用于日常生产和生活。VoIP语音通信技术不仅信息传输质量较好,还满足了目前的语音通信需要,对于语音通信技术的进一步发展有着积极的促进作用。基于VoIP语音通信技术进行研究与问题分析,以供参考。展开更多
为提高基于IP的语音传输(Voice over Internet Protocol,VoIP)音频通信系统的可用性,设计并实现了一款基于会话初始协议(Session Initiation Protocol,SIP)的高可用VoIP音频通信系统。首先,基于虚拟IP技术,建立系统在路由层的高可用框...为提高基于IP的语音传输(Voice over Internet Protocol,VoIP)音频通信系统的可用性,设计并实现了一款基于会话初始协议(Session Initiation Protocol,SIP)的高可用VoIP音频通信系统。首先,基于虚拟IP技术,建立系统在路由层的高可用框架。其次,在SIP协议栈层搭建SIP服务器集群,实现系统在网络音频通信业务层的高可用和负载均衡。最后,使用关系型数据库,成功搭建集群在数据一致性方面的高可用模型。实验结果表明,通过该设计可以有效降低VoIP通信系统的故障停机时间,同时可以提高系统的业务负载能力。展开更多
IP的语音传输(Voice over Internet Protocol,VoIP)技术是一种基于网络的语音技术,它可以将语音信号转换成数据流,并在互联网上传输。VoIP技术与Call Center的融合使用户在通话时能更加方便、快捷地使用互联网,节省时间和费用,实现语音...IP的语音传输(Voice over Internet Protocol,VoIP)技术是一种基于网络的语音技术,它可以将语音信号转换成数据流,并在互联网上传输。VoIP技术与Call Center的融合使用户在通话时能更加方便、快捷地使用互联网,节省时间和费用,实现语音信号与数据流的同时传输,提高通话质量和效率。同时,技术融合可以解决传统呼叫中心面临的一些挑战,如服务质量下降、成本上升等问题。展开更多
文摘Because of the best-effort service in Internet, direct routing path of Internet may not always meet the VoIP quality requirements. Thus, many researches proposed Peer-to-Peer VoIP systems such as SIP+P2P system, which uses relay node to relay RTP stream from the source node to the destination node and uses application-layer routing scheme to lookup the best relay nodes. The key of those systems is how to lookup the appropriate relay nodes, which we call relay lookup problem. This paper presents a novel peer relay lookup scheme based on SIP+P2P system. The main ideas are to organize the P2P network using a Cluster overlay and to use topology-aware to optimize relay selection. We introduce the mechanism in detail, and then evaluate this mechanism in NS2 network simulation environment. The results show that our scheme is scalable and can get high relay hit ratio, which confirm the feasibility of a real system. We also make comparison with traditional schemes and the results show that our scheme has good path quality.
文摘针对集中式基于网际互连协议的语音传输(Voice over Internet Protocol,VoIP)语音通信交换系统(Voice Communication Switching System,VCCS)的互操作性需求,提出一种基于VoIP技术的中继通信解决方案。在集中式VoIP通信架构的基础上构建中继通信模型,结合局向管理、冠号管理、路由管理以及会议管理等模块的协同运行给出VoIP中继通信的业务处理流程,并详细阐述中继会议的核心模块。最后,通过两种异型语音通信交换系统的中继通信对比测试,验证了文章提出的VoIP中继通信方法的可靠性和互操作性。
文摘VoIP (Voice over IP) is a rapidly growing area with great market potential. To promote it for both commercial and research purposes, a prototype VoIP system based on state-of-the-art Motorola communication techniques has been developed. It is a gateway system integrating a PBX and a VoIP module. All components that H.323 defines to support VoIP are implemented in the VoIP module, though in a simplified manner. As an embedded system, the system features real timeness and task distributiveness. A number of additional techniques are used to improve the performance, including noise suppression, zero copy, and buffer structure optimization. When refined in interoperability, the system will also readily serve as a product.
文摘In this paper, the capability of IEEE 802.11b Distributed Coordination Function (DCF) mode supporting of Constant Bit Rate (CBR) and Variable Bit Rate (VBR) Voice over IP (VoIP) traffic is investigated. Then, the capacity of 802. lib Wireless Local Area Network (WLAN) system carrying zoice calls in a wide range of scenarios, including varying delay and packet loss rate constraints is analyzed and evaluated. Both G. 711 and G. 729 voice encoding schemes and a range of voice inter-arrival time are considered. The analyses and simulation results show that capacity is highly sensitive to the delay budget allocated to the sum of pucketization and wireless netzoork delays. For a given packet loss rate constrained, G. 729 is shown to have a capacity greater than that when G. 711 is used. The simulation results show that by supporting VBR under DCF mode the network has the approximately twice much capacity as supporting CBR has, regardless of the encoding schemes and the inter-arrival time.
文摘随着互联网技术的进步,语音通信也经过了技术变革。基于IP的语音传输(Voice over Internet Protocol,VoIP)技术在近年来成为社会关注的热点话题,逐渐广泛应用于日常生产和生活。VoIP语音通信技术不仅信息传输质量较好,还满足了目前的语音通信需要,对于语音通信技术的进一步发展有着积极的促进作用。基于VoIP语音通信技术进行研究与问题分析,以供参考。
文摘为提高基于IP的语音传输(Voice over Internet Protocol,VoIP)音频通信系统的可用性,设计并实现了一款基于会话初始协议(Session Initiation Protocol,SIP)的高可用VoIP音频通信系统。首先,基于虚拟IP技术,建立系统在路由层的高可用框架。其次,在SIP协议栈层搭建SIP服务器集群,实现系统在网络音频通信业务层的高可用和负载均衡。最后,使用关系型数据库,成功搭建集群在数据一致性方面的高可用模型。实验结果表明,通过该设计可以有效降低VoIP通信系统的故障停机时间,同时可以提高系统的业务负载能力。
文摘IP的语音传输(Voice over Internet Protocol,VoIP)技术是一种基于网络的语音技术,它可以将语音信号转换成数据流,并在互联网上传输。VoIP技术与Call Center的融合使用户在通话时能更加方便、快捷地使用互联网,节省时间和费用,实现语音信号与数据流的同时传输,提高通话质量和效率。同时,技术融合可以解决传统呼叫中心面临的一些挑战,如服务质量下降、成本上升等问题。