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VoIP over the WiMAX

VoIP over the WiMAX. Adviser: Ho-Ting Wu Presenter: Chi-Fon Yang. Outline. VOIP Protocols IEEE 802.16 Introduction Voice over Ethernet via IEEE 802.16 QoS Strategy for VoIP Services over IEEE 802.16 Conclusions References.

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VoIP over the WiMAX

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  1. VoIP over the WiMAX Adviser: Ho-Ting Wu Presenter: Chi-Fon Yang

  2. Outline • VOIP Protocols • IEEE 802.16 Introduction • Voice over Ethernet via IEEE 802.16 • QoS Strategy for VoIP Services over IEEE 802.16 • Conclusions • References

  3. 主要的功能包括:遠端自動啟用安全編碼,HTTP,TFTP & FTP電話功能:多條通話線(最多4個sip帳戶)記憶撥號(最多9組)免持聽筒電話預撥 聽筒和免持聽統音量調節速撥(10組)電話簿(200筆記錄)多條線路(最多10條)通話記錄(來話/去話/未接)支援SIP ServerRegistrar(網頁設定)Outbound proxy(網頁設定)撥號方式:直接IP Address撥號無需在SIP Server上註冊撥在SIP Server註冊的號碼從電話簿/速撥鍵撥出 聲音品質:VAD聲音偵測CNG消除雜音AEC G.168消除回音Jitter Buffer延遲補償 VoIP PCI card for Asterisk IP PBX server WiFi電話-雙網手機 美國盛行的ATA網路電話盒 Outline • VOIP Protocols • Uses the Internet Protocol (IP) to transmit voice as packets over an IP network. • The main focus is on H.323 and SIP (Session Initiation Protocol) . NEWS

  4. VOIP Protocols • Signaling protocols • H.323 , SIP are used to setup the route for the transmission over the IP network • Signaling transport (SIGTRAN) are used transport SS7 over IP • Gateway protocols • Media Gateway Control Protocol (MGCP/MEGACO) are used to establish control and status in the media and signaling gateways. • Routing (UDP,TCP) and transport protocols (RTP) are used once the route is established for the transport of the data stream

  5. H.323 • ITU-T standard • Provides the technical requirements for voice communication over IP service • Provides audio, video and data communications across IP-based networks. • Control protocols • H.225.0/Q.931 Call Signaling • H.225.0 RAS • H.225.0 Call Signaling • H.245 Media Control

  6. Components of H.323 • Terminal • The LAN client endpoints that provide real time • Support H.245, Q.931, RAS, RTP,MCU • Gateways • Two-way communications between H.323 terminals • Translation between different transmission formats (e.g from H.225 to H.221 ) • Translating between audio and video codecs • Gatekeepers • Address Translation • Admissions Control • Call signaling • Bandwidth Management • Call Management • Multipoint Control Units (MCU) • Provides the capability for three or more terminals and gateways to participate in a multipoint conference Call-Setup

  7. SIP messages • SIP-message = Request | Response • The request specifies the type of request • The response indicates the success or failure of a given request start-line = request-Line | status-Line Generic-message = start-line *message-header CRLF [message-body] message-header = (general-header | request-header | response-header | entity-header) SIP - Session Initiation Protocol • IETF standard for VOIP • RFC 3261 from the Internet Engineering Task Force (IETF) • Application layer control protocol for creating, modifying and terminating sessions • Similar to that of HTTP or SMTP • Text-encoded protocol • Main functions ‧Invite users to sessions • Find the user’s current location, match with their capabilities and preferences ‧Modification of sessions ‧Termination of sessionsText-based Encoding

  8. SIP Architecture SIP Request SIP Response RTP Media Stream Redirect Server Location Server Proxy Server Proxy Server Proxy Server

  9. COMPARISON OF H.323 WITH SIP

  10. Media Gateway Control Protocol MGCP/MEGACO • Defines communication between call control elements (Call Agents) and telephony gateways.

  11. MGCP/MEGACO Introduction • A protocol for controlling media gateways • Components • Media Gateway (MG) – • provides conversion between audio signals on telephone circuits and data packets carried over IP networks. • Trunking gateway – • to CO/switches • Residential gateway – • to PSTN phones • Media gateway controller (MGC) – • handles the call setup and release for media channels in a media gateway. Call-Setup

  12. MGC IMS Softswitch MG

  13. Outline • VOIP Protocols • IEEE 802.16 Introduction

  14. IEEE 802.16 Introduction • FEATURES OF WIMAX • Scalability : • The standard supports hundreds or even thousands of users within one RF channel • As the number of subscribers grow the spectrum can be reallocated with process of sectoring • Quality of Service • Range : • Optimized for up to 50 Km • Designed to handle many users spread out over kilometres • Coverage : • Standard supports mesh network topology • Downlink Channel • Downlink Channel Descriptor (DCD) • Transmitted by the BS • Define the characteristics of a downlink physical channel • Downlink Map • PHY Synchronization • DCD Count • Match the DCD • Base Station ID • 48-bit • SBC-REQ / SBC-RSP management message – • SBC-REQ stands for SS Basic Capabilities Request. • SBC-RSP stands for SS Basic Capabilities Response. • Negotiate basic capabilities. • The SS informs the BS of its basic capabilities • IEEE 802.16 • Worldwide interoperability of Microwave Access High demand for “last-mile” broadband access. • Provide hight-speed internet access to home and business subscribers, without wires • Authorize/key exchange – • For authorization procedure • PKM-REQ /PKM-RSP management message – • PKM ( Privacy Key Management ) Protocol • Uplink Channel • Uplink Channel Descriptor (UCD) • Configuration Change Count • Determine whether the content is changed • Mini-Slot Size • Units of Physical Slot, Allowable n=2m • Uplink Channel ID • Arbitrarily Chosen by the BS • Unique within the MAC-Sublayer domain • Uplink Map (UL-MAP) • Allocates access to the uplink channel • REG-REQ / REG-RSP management message – • Initialization registration procedure • The process by which the SS is allowed entry into • network and becomes manageable

  15. Frame structure of IEEE 802.16

  16. 9. Configuration information Downlink channel Step to Network Access parameter 7. Management msg SS parameter BS 1. SCAN for a downlink channel 2. DL-MAP 3.DCD 4. Adjust local parameter 8. Establish time of day 6.Authorize/key exchange 5. SBC-REQ 7. REG-REQ

  17. Downlink channel Uplink channel Step to Channel Access SS BS DATA Time slot DATA

  18. Outline • VOIP Protocols • IEEE 802.16 Introduction • Voice over Ethernet via IEEE 802.16

  19. Voice over Ethernet via IEEE 802.16 • The new business opportunities created by the advance of new technologies -less expensive than 3G technologies (e.g., Wi-fi, WiMAX). • VoIP is considered as a compulsory service (free or not-free of charge) in addition to any other services. • Compared with traditional circuit-switched telephone system, many people believe “voice over packet-switched networks has the potential of delivering a more cost-effective service solution.”

  20. Voice over Ethernet via IEEE 802.16 • Downlink • The Base Station (BS) packs the incoming VoIP packets for several users into a downlink (DL) – burst. • Transmits burst to the Subscriber Station (SS) in a single DL-subframe. • The SS extracts its packets and forwards them to the VoIP applications in the Ethernet. • Uplink • upstream VoIP packets into an uplink (UL) – burst. • Transmits them to the BS in a single UL-subframe. • The BS then extracts the original VoIP packets and forwards them to the Internet

  21. Transmission Time of a fixed-size frame • T is the transmission time of a fixed-size frame. • P is the transmission time of the preambles, • B is the transmission time of broadcast messages • RC is the register contention time. • (V + MAC) is the transmission time of a voice payload with MAC header • C be the maximum number of VoIP sessions that can be supported by IEEE 802.16 point to-point connection (C ≈ 60).

  22. VoIP characteristics and requirements • VoIP traffic • Strict delay latency requirement (RF delay 70 ms in simulation) • Multiple VOIP user multiplexing is critical • OFDMA: 32 users in every 5 ms per sector • HSDPA: 4 users in every 2 ms per sector • DO-A: 8 users in every 1.667 ms per sector • Requirements for Voice • Typically target a loss rate of 0.25 percent or less. • One-way latency should be no more than 150 ms. • Jitter should be less than 10 ms.

  23. Outline • VOIP Protocols • IEEE 802.16 Introduction • Voice over Ethernet via IEEE 802.16 • QoS Strategy for VoIP Services over IEEE 802.16

  24. QoS Strategy for VoIP Services over IEEE 802.16 • Qos ( Quality of Service ) • Associate a packet with a service flow. • Service Flow：unidirectional flow of packets that provides a particular Qos. • Support Quality of Service • Unsolicited Grant Service - UGS • Real-Time Polling Service – rtPS (Real-Time Polling Service) • Non-Real-Time Polling Service – nrtPs • Best Effort – BE • UGS - • Real-time service flows • Periodic, fixed size grants • Avoid overhead and latency of frequent SS redundant • Meet the continuous need of service flow • T1 / E1 / VoIP • rtPS - • Real-time service flows • Variable size data • MPEG • nrtPs - • Non real-time service flow • Variable size data • Best Effort – BE • Efficient service to best effort

  25. QoS functions within the BS and SSs

  26. The Effect of factors on Speech Quality • Loss – • Comparative measure of packets transmitted and received to the total number of packets that were transmitted. • Delay – • The finite amount of time it takes a packet to reach the receiving endpoint after being transmitted from the sending endpoint. • Jitter – • The difference in the end-to-end delay between packets. • Throughput – • The available user bandwidth between an ingress point and an egress point • Voice transmission over wireless brings along with a it a big problem of packet delay or latency.

  27. VoiceQuality computational scheme • R-factor – • The E-model produces a single value • Derived from a variety of factors, eg. Delay • The range from 0 (extremely poor) to 100 (high quality) • Any R-factor below 50 is unacceptable • Three main variations of R-factor • Call quality estimate, RCQE . • Listening quality estimate , RLQE . • Network performance estimate , RNPE . • E-Model - • Provides a prediction of the expected voice quality • Originally the E-Model was intended for use in network planning and design.

  28. R-factor • In the E-Model several different parameters affecting the quality of a conversation are taken into account. • R0 is the basic signal-to-noise ratio (environmental and device noises). • Is accounts for the impairments on the coded voice signal (loud connection and quantization) • Id represents the effect of delay • Ie the effect of low bit rate codecs • A is the advantage factor, corresponding to the user allowance due to the convenience in using a given technology.

  29. Committed Information Rate and Maximum Information Rate • Two main parameters are used in order to support service differentiation at the higher layers: • Rmax - the maximum traffic rate available at the WiMAX Downlink Air Interface. • CIRk and MIRk - the request of the k-th SS2. • Committed Information Rate (CIR) – • The CIR parameter for a WiMAX system is the bit-rate that the network agrees to accept from the user. • Maximum Information Rate (MIR) – • regulates the maximum allowed peak rate of a connection. MIR and CIR are specified for each SS according to the negotiated Service Level Agreement (SLA)

  30. VOIP service flow allocation • The opposite case – • when the aggregate of the CIR requested by VoIP sub-scribers exceeds Rmax • RBE (bit/s) - BE Service Rate. • BS can provide to the m-th TCP service flow • RTCP(m) be the service rate that the BS can provide to the m-th TCP service flow. • RRT (bit/s) - Real Time (RT) Service Rate. • usual assumption is that the BE flows are TCP friendly • RVoIP(m) is the service rate that the BS provides to the m-th VoIP service flow • Ntot - total number of downstream service flows • consisting of NVoIP(VoIP flows) and NTCP(TCP persistent connection). • WiMAX network is deterministically lower than Rmax (no congestion occurs). 1. 將的資源優先分配給較高property的 voice service flow 2. 若在BS busy的狀態下 ，VOIP signaling (BE property的 service)無法搶得BS資源. 3. BE service的property的最低， RTCP使用RVoIP剩下的資源後的BS資源

  31. Performance evaluationof WiMAX under VoIP Traffic • Deployed in Turin, Italy, within the national experimentation on WiMAX coordinated by Fondazione Ugo Bordoni (http://wimax.fub.it). • The distance between the BS and SS1, SS2 and SS3 is 8.4 km, 8.5 km and 13.7 km, respectively. • The equipment operates using a 3.5 MHz wide channel in FDD mode, The SSs work in under FDD half-duplex mode. 64 QAM, for each connection. • The average signal-to-noise ratio is above 30 dB. • All nodes run Linux with a 2.4 kernel.

  32. Performance evaluationof WiMAX under VoIP Traffic • VoIP codecs feed RTP packet flows and three commonly used codecs (G.711,G.729.2 and G.723.1) • rtPS services are used forVoIP connections. • TCP-controlled traffic is mapped in the BE class.

  33. Packet loss rate of VoIP flows per SS using different codecs

  34. Average Delay vs Number of SS VoIP flows G.729.2 packet generation rate, coupled to the large overhead of packet headers of the RTP/UDP/IP/MAC protocol stack (45% for the G.729.2).

  35. Average Throughput per VoIP session versus an increasing number of VoIP per SS

  36. Average R-factor versus number of SS VoIP flows

  37. Packet Delay pdf

  38. Outline • VOIP Protocols • IEEE 802.16 Introduction • Voice over Ethernet via IEEE 802.16 • QoS Strategy for VoIP Services over IEEE 802.16 • Conclusions

  39. Conclusions • We discussed the transmission of VoIP and IMS over WiMAX. • The working of a Softswitch which makes VoIP over WiMAXpossible. • WiMAX is to ensure large area coverage and rather inexpensive equipment at the subscriber side. Modern requirements to wireless connectivity include mandatory QoS guarantees for a wide set of real-time applications. • Wireless Service Providers (WSPs) use WiMAX networks to provide connectivity to both residential (voice, data and video) and business (primarily voice and Internet) customers.

  40. Outline • VOIP Protocols • IEEE 802.16 Introduction • Voice over Ethernet via IEEE 802.16 • QoS Strategy for VoIP Services over IEEE 802.16 • Conclusions • References

  41. References • [WiMAX05] WiMAX Forum: "Can WiMAX Address your Applications?“ • http://www.wimaxforum.org/news/downloads/Can_WiMAX_Address_Your_Applications_final.pdf • [JDSU03]JDSU:"White Paper – VoIP Overview"-JDSU Corporation 2003 • http://www.jdsu.com/test_and_measurement/technical_resources/product_documents/whitepaper/voipterm_wp_acc_tm_ae_1205.pdf • Performance Evaluation of a WiMAX Testbed under VoIP Traffic http://portal.acm.org/citation.cfm?id=1161009&jmp=cit&coll=GUIDE&dl=G&CFID=15151515&CFTOKEN=6184618 • C. Cicconetti, L. Lenzini, E. Mingozzi, and C. Eklund.Quality of service support in IEEE 802.16 networks. IEEE Network Magazine, 20(2):50–55, March 2006. • F. De Pellegrini, D. Miorandi, E. Salvadori and N. Scalabrino.QoS Support in WiMAX Networks: Issues and Experimental Measurements. Technical Report 200600009, CREATE-NET, 2006. • C. Hoene, H. Karl, and A. Wolisz. A perceptual quality model intended for adaptive VoIP applications: Research articles. Int. J. Commun. Syst., 19(3):299–316, 2006.

  42. Q and A THANK YOU

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