Voice over IP phone Software Design. Software Engineering Project University of Oulu 2004
|
|
- Darlene Webster
- 8 years ago
- Views:
Transcription
1 Voice over IP phone Software Design Software Engineering Project University of Oulu 2004
2 Page 2 of 48 Table of Contents 1. Version Terminology...5 Abbreviations...5 Terms Procedural Design...6 General...6 Process and threads...6 Internal messages of process...7 Comments / restrictions / limitations Test Provisions...9 Test guidelines...9 Module testing...9 Interface testing...9 Special considerations Scope...11 Major Software requirements...11 Design constraints, limitations Human to machine interface specification...12 RS232 interface...12 Requirements...12 implementation...12 LCD-display...16 Requirements...16 implementation...16 LEDs...17 Requirements...17 implementation...17 Speaker...19 Requirements...19 implementation...19 Microphone...19 Requirements...19 implementation Data structures...21 External / interface data structures...21 RTP header...21 RTP packet...21 Internal data structures...21 Network configuration...21 IP address and port...22 User configuration...22
3 Page 3 of 48 SIP connection information...22 Common RTP packet data...22 LCD text Notes References Appendix...25
4 Page 4 of Version Version v Table 1Version Date(dd.mm.yyyy) Author Comments First version
5 Page 5 of 48 2.Terminology Abbreviations Abbreviation SIP ARP RTP ICMP UDP RFC Explanation Session Initiation Protocol Address Resolution Protocol Real-time Transport Protocol Internet Control Message Protocol User Datagram Protocol Request for Comments Table 2Abbreviations Terms Term Stubs RFC Explanation A stub is something which is short, underdeveloped, or partially cut off. Internet informational documents and standards Table 3Terms
6 Page 6 of 48 3.Procedural Design General VoIP phone will be implemented as one process main and multiple light processes called threads. Used OS is NutOS which is rather feature rich OS designed especially for Ethernut HW. Main idea of our SW design is to cover every individual HW resource (rs232, lcd, led, microphone, etc.) behind one thread. Variables (constants, data structures) can be accessed directly only one tread. With this arrangement we can avoid critical section problems and possible very hard to find bugs. If tread needs data owned by other tread then it must ask it by using XgmessageQue. All treads have own message queue where it check if new message is arrived. Process and threads Process / thread Main LCD LED Mic Spkr RTP SIP RS232 Task Head of the family. Maintains phone common settings. Controls LCD. Controls LEDs. Controls microphone. Controls speaker. Controls RPT session. Handles RTP packet sending / receiving. Encode / decode voice. Controls SIP sessions. Handles SIP packet sending / receiving. Listens SIP port for incoming connections. Controls serial interface. Handles serial UI. Table 4Process structure
7 Page 7 of 48 Internal messages of process Message Name id LCD_WRITE 101 LCD_CLEAR 102 LED_ON 201 LED_OFF 202 LED_BLINK 203 SPKR_ON 301 SPKR_OFF 302 SPKR_ALERT 303 SPKR_BUSY 304 SPKR_RINGBACK 305 MIC_ON 401 MIC_OFF 402 INCOMING_INVITE 501 OUTCOMING_INVITE 502 INCOMING_BUSY_HERE 503 OUTCOMING_BUSY_HER 504 E INCOMING_ACK 505 OUTCOMIG_ACK 506 INCOMING_BYE 507 OUTCOMING_BYE 508 INCOMING_OK 509 OUTCOMING_OK 510 INCOMING_RINGING 511 OUTCOMING_RINGING 512 INCOMING_USER_NOT_F 513 OUND OUTCOMING_USER_NOT 514 _FOUND INCOMING_REQUEST_TE 515 RMINATE OUTCOMING_REQUEST_ 516 TERMINATE CALL 601 END_CALL 602 ANSWER_CALL 603 RTP_CONNECT 701 RTP_DISCONNECT 702 Table 5Messages from to Parameter data Information line parameter 0 writes lcd lcd_text_t number all rows lcd led 1 to 8 0 turn off all LEDs led 1 to 8 led 1 to 8 spkr connect spkr disconnect spkr spkr spkr mic connect mic disconnect sip main sip_user_t main sip sip_user_t sip main sip_user_t main sip sip_user_t sip main main sip sip main main sip sip main main sip sip main main sip sip_user_t sip_user_t sip_user_t sip_user_t sip_user_t sip_user_t sip_user_t sip_user_t sip sip_user_t main main sip sip_user_t sip sip_user_t main main sip sip_user_t rs232main rs232main rs232main main rtp main rtp sip_user_t sip_user_t iaddr_t
8 Comments / restrictions / limitations arp and icmp echo reply is provided by OS. enough CPU and other HW resources? does messagequeue works as expected? NutOS limitations? Possible OS bugs? how many threads is possible to use? how much memory one threads takes? scheduling issues between threads and master? Page 8 of 48
9 Page 9 of 48 4.Test Provisions Testing is done in two phases; module- and interface testing. The aim of module testing is ensure internal functionality of our process family and aim of interface testing is verify protocol messaging. Module testing will be executed against software stubs and interface testing between two terminal equipments. Test guidelines Module testing Tools: software stubs network analyzer debugger test logs Things to be tested messaging between treads timers error handling user interface sip, rtp, (arp, icmp) Interface testing Tools: network analyzer other terminal equipment (IP-phone) Things to be tested
10 messaging between terminal equipments voice encoding/decoding voice transmission Page 10 of 48 Special considerations Speech quality can be measured by recording/hearing received voice and frequency response by example generating needed tones using signal generator (sin-signal) and measuring received signal by using oscilloscope. Also it could be possible to capture/record voice via computer sound-card and then measure signals using some audio software.
11 Page 11 of 48 5.Scope Major Software requirements Work assignment gives following guidelines: The VoIP phone has similar functionality as we know it from our contemporary telephone: It alerts the callee, user can make a call, transmits voice, receives voice and terminates a call. The phone uses state of the art Internet protocols: SIP (Session Initiation Protocol) and RTP (Real Time Protocol). SIP serves as the control protocol - to initiate and terminate calls. RTP is transporting the encoded voice. [Course Assignment; A Software Engineering Project] Work assignment can be reached with parallel processing and distributing tasks to multiple threads. Parallel processing is required to get system working smoothly. Asynchronous data exchange between processes and threads will be used. NutOS is rather versatile real time platform which offers many helpful functionalities for multitasking. Process with multiple threads can be used to meet real time requirements. Only RTP data transfer needs quite hard real time processing. Design constraints, limitations See chapter: Comments / restrictions / limitations
12 Page 12 of 48 6.Human to machine interface specification User interfaces are: RS232 terminal connection (input/output) 4*16 characters text mode LCD-display (output) 8*LED (output) Speaker Microphone Ethernut board includes RS232 port and LEDs, LCD-display, speaker and microphone found on external IO-board which is connected to Ethernut board. RS232 interface Requirements call answer hangup reject on-line help set user set IP address set IP hostmask set MAC address (HW address) show configuration implementation startup (power on): Connected... NutOS version x.y.z
13 Page 13 of 48 Use help for basic help. Use help <command> for help on a specific command. help command: prompt> help List of available commands: call answer hangup help set_user set_ip set_hostmask set_mac show_config help call command: prompt> help call ### call <sip url (sip:username@host)> Calls to given user. Host can be given only in number format. For example 'call sip:babo@ ' help answer command: prompt> help answer ### answer Answers to incoming call. Answer command can be shortened as one charter 'a'. help hangup command: prompt> help hangup ### hangup Disconnects ongoing call or rejects incoming call depending which one situation is in case. Hangup command can be shortened as one charter 'h'.
14 help help command: prompt> help help ### help [command] Help command without any arguments prints out the list of all possible commands. Use help <command> for help on a specific command. help set_user command: prompt> help set_user ### set_user <username> Set/change phone user. Username length can be between 1-xx characters. help set_ip command: prompt> help set_ip ### set_ip <x.x.x.x> Sets/change phone ip address. Every decimal must be between IPv6 addressing is not supported. help set_hostmask command: prompt> help set_hostmask ### set_hostmask <x.x.x.x> Sets/change phone hostmask. Every decimal must be between IPv6 addressing is not supported. help set_mac command: prompt> help set_mac ### set_mac <XX:XX:XX:XX:XX:XX> Sets/change phone ethernet HWaddress. Every value must be between 0-FF (HEX values). help show_config command: Page 14 of 48
15 Page 15 of 48 prompt> help show_config ### show_config Outputs current confguration. (user, ip address, hostmask, mac address) show_config command: prompt> show_config Current configuration: user: john ip: hostmask: mac: 00:0D:88:C8:24:76 execution, command success: OK: command executed obs. this text is not implemented at the first phase. It will be added later if necessary. execution error, unknown command: ERROR: unknown command execution error, invalid value (user enters unreasonable values): ERROR: invalid value execution error, change is not possible due to ongoing call etc.: ERROR: change not possible at the moment $ (dollar + one space) will be used as prompt. Enter without any
16 Page 16 of 48 character (empty command) prints new prompt on new line (linefeed + prompt). LCD-display Requirements display counterpart SIP address display call information display call duration (will be implemented if enough time) implementation Outcoming call: Call to Incoming call: Call from sip:us er@host Call connected: Call connected t o sip:user@host Call disconnected: Call disconnecte d by sip:user@ho st
17 Call canceled: Call canceled by User not found: User not found s ip:user@host User busy: User busy sip:us er@host Idle state: Idle LEDs Requirements display phone state implementation Outcoming call: LED 1 blinking Incoming call: LED 2 blinking Page 17 of 48
18 Outcoming call connected: LED 1 on Incoming call connected: LED 2 on Call disconnected: LEDs off Call canceled by local end: LED 3 on Call canceled by remote end: LED 4 on Local user not found: LED 7 on Remote user not found: LED 8 on Local user busy: LED 5 blinking Remote user busy: LED 6 on Idle state: LEDs off Page 18 of 48
19 Page 19 of 48 Speaker Requirements alerting voice implementation Incoming call: Alerting tone. Typical ringing tone; to be defined better during implementation. Remote user busy: Busy tone. Waiting for remote answer: Ring-Back tone. Call connected: Remote users voice. Dial tone is not needed and not implemented. Microphone Requirements Receive voice
20 implementation Not relevant. Page 20 of 48
21 7.Data structures External / interface data structures RTP header typedef struct { unsigned int version:2; unsigned int p:1; unsigned int x:1; unsigned int cc:4; unsigned int m:1; unsigned int pt:7; u_int16 seq; u_int32 ts; u_int32 ssrc; u_int32 csrc[1]; } rtp_hdr_t; /* /* /* /* /* /* /* /* /* /* protocol version */ padding flag */ header extension flag */ CSRC count */ marker bit */ payload type */ sequence number */ timestamp */ synchronization source */ optional CSRC list */ [12] Speak-Freely, RTP packet #define MAX_PAY xxx typedef struct { rtp_hdr_t hdr; /* RTP headers */ u_char payload[max_pay]; /* RTP payload */ } rtp_packet_t; Internal data structures Network configuration typedef struct { u_long iaddr; u_long mask; u_long gateway; u_char hwaddr[16]; } ethconf_t; /* /* /* /* IP address */ hostmask */ gateway IP address */ Hardware address */ Page 21 of 48
22 Page 22 of 48 IP address and port typedef struct { u_long remote_iaddr; u_short remote_port; } iaddr_t; /* remote IP */ /* remote port */ User configuration #define USER_LEN xxx #define HOST_LEN xxx typedef struct { char user[user_len]; char host[host_len]; } userconf_t; /* username */ /* host */ SIP connection information #define SURI_LEN xxx typedef struct { char from[suri_len]; char to[suri_len]; char cseq[32]; iaddr_t remote_addr } sip_user_t; /* /* /* /* SIP uri */ SIP uri */ call sequence */ remote IP / port */ Common RTP packet data typedef struct { unsigned int version:2; unsigned int p:1; unsigned int count:5; unsigned int pt:8; u_int16 length; word */ } rtcp_common_t; /* /* /* /* /* protocol version */ padding flag */ varies by payload type */ payload type */ packet length in words, without this [12] Speak-Freely, LCD text typedef union { char all[64]; char row[4][16]; } lcd_text_t; /* whole text */ /* text in a rows */
23 Page 23 of 48 8.Notes Some parts software design is defective due to lack of time. Course assignment is too large to design complete given time. Major weakness of our design is missing design of treads. Missing design will be updated during this project.
24 Page 24 of 48 9.References [1] Course Assignment; A Software Engineering Project, [2] RFC3261, SIP: Session Initiation Protocol, [3] RFC2327, SDP: Session Description Protocol, [4] RFC3665, Session Initiation Protocol (SIP) Basic Call Flow Examples, [5] RFC1889, RTP: A Transport Protocol for Real-Time Applications, [6] RFC1890, RTP Profile for Audio and Video Conferences with Minimal Control, [7] The Internet Encyclopedia, G.711 Protocol Overview, [8] Understanding SIP, [9] Functions for g711 coding, udioresources/g.723/g711.c [10] Ethernut HW and NutOS manufacturer, [11] Telecommunications Reference Index, [12] Speak-Freely,
25 10.Appendix All diagrams are developed using Prosa UML tools. Use case diagram use_case.ucd (hw) Class diagram class_diagram.cld(hw) Sequence diagram call.sqd (hw) other_call.sqd (hw) disconnect.sqd (hw) other_disconnect.sqd (hw) not_found.sqd (hw) other_not_found.sqd (hw) busy.sqd (hw) other_busy.sqd (hw) terminate.sqd (hw) other_terminate.sqd (hw) ethernut_call.sqd (sw) ethernut_other_call.sqd (sw) ethernut_disconnect.sqd (sw) ethernut_other_disconnect.sqd (sw) ethernut_not_found.sqd (sw) ethernut_other_not_found.sqd (sw) ethernut_busy.sqd (sw) ethernut_other_busy.sqd (sw) ethernut_terminate.sqd (sw) ethernut_other_terminate.sqd (sw) Page 25 of 48
26 Date and place Oulu Antti Palosaari Jaakko Tiri Kimmo Hettula Page 26 of 48
27 Page 27 of 48
28 Page 28 of 48
29 Page 29 of 48
30 Page 30 of 48
31 Page 31 of 48
32 Page 32 of 48
33 Page 33 of 48
34 Page 34 of 48
35 Page 35 of 48
36 Page 36 of 48
37 Page 37 of 48
38 Page 38 of 48
39 Page 39 of 48
40 Page 40 of 48
41 Page 41 of 48
42 Page 42 of 48
43 Page 43 of 48
44 Page 44 of 48
45 Page 45 of 48
46 Page 46 of 48
47 Page 47 of 48
48 Page 48 of 48
point to point and point to multi point calls over IP
Helsinki University of Technology Department of Electrical and Communications Engineering Jarkko Kneckt point to point and point to multi point calls over IP Helsinki 27.11.2001 Supervisor: Instructor:
More informationP160S SIP Phone Quick User Guide
P160S SIP Phone Quick User Guide Version 2.2 TABLE OF CONTENTS 1.0 INTRODUCTION... 1 2.0 PACKAGE CONTENT... 1 3.0 LIST OF FIGURES... 2 4.0 SUMMARY OF KEY FUNCTIONS... 3 5.0 CONNECTING THE IP PHONE... 4
More informationTransport and Network Layer
Transport and Network Layer 1 Introduction Responsible for moving messages from end-to-end in a network Closely tied together TCP/IP: most commonly used protocol o Used in Internet o Compatible with a
More informationRTP / RTCP. Announcements. Today s Lecture. RTP Info RTP (RFC 3550) I. Final Exam study guide online. Signup for project demos
Announcements I. Final Exam study guide online RTP / RTCP Internet Protocols CSC / ECE 573 Fall, 2005 N. C. State University II. III. Signup for project demos Teaching evaluations at end today copyright
More informationIP-Telephony Real-Time & Multimedia Protocols
IP-Telephony Real-Time & Multimedia Protocols Bernard Hammer Siemens AG, Munich Siemens AG 2001 1 Presentation Outline Media Transport RTP Stream Control RTCP RTSP Stream Description SDP 2 Real-Time Protocol
More informationProvisioning and configuring the SIP Spider
Provisioning and configuring the SIP Spider Administrator Guide Table of Contents 1. Introduction... 3 2. Manual Provisioning... 4 3. Automatic Provisioning... 5 3.1 Concept... 5 3.2 Preparing the configuration
More informationNAT TCP SIP ALG Support
The feature allows embedded messages of the Session Initiation Protocol (SIP) passing through a device that is configured with Network Address Translation (NAT) to be translated and encoded back to the
More informationDPH-140S SIP Phone Quick User Guide
DPH-140S SIP Phone Quick User Guide Version 1.0 TABLE OF CONTENTS 1.0 INTRODUCTION... 1 2.0 PACKAGE CONTENT... 1 3.0 LIST OF FIGURES... 2 4.0 SUMMARY OF KEY FUNCTIONS... 3 5.0 CONNECTING THE IP PHONE...
More informationV101 SIP VoIP Telephone Adaptor User Manual V1.1m
V101 SIP VoIP Telephone Adaptor User Manual V1.1m Quick Guide Step 1: Broadband (ADSL/Cable Modem) Connections for V101 A. Connect V101 LAN port to ADSL NAT Router as the following connection. B. Connect
More informationKapanga The Media over IP Softphone. Quick Start Manual April 2005
Kapanga The Media over IP Softphone Quick Start Manual April 2005 Quick Start Manual This manual briefly describes the interface, menus and settings available to a Kapanga user. While this document assumes
More informationAdvanced Networking Voice over IP: RTP/RTCP The transport layer
Advanced Networking Voice over IP: RTP/RTCP The transport layer Renato Lo Cigno Requirements For Real-Time Transmission Need to emulate conventional telephone system Isochronous output timing same with
More informationVoice over IP: RTP/RTCP The transport layer
Advanced Networking Voice over IP: /RTCP The transport layer Renato Lo Cigno Requirements For Real-Time Transmission Need to emulate conventional telephone system Isochronous output timing same with input
More informationAculab digital network access cards
Aculab digital network access cards Adding and Using IPv6 Capabilities Guide Revision 1.0.2 PROPRIETARY INFORMATION Aculab Plc makes every effort to ensure that the information in this document is correct
More informationWelcome. Unleash Your Phone
User Manual Welcome Unleash Your Phone For assistance with installation or troubleshooting common problems, please refer to this User Manual or Quick Installation Guide. Please visit www.vonage.com/vta
More informationInteroperability Test Plan for International Voice services (Release 6) May 2014
INTERNATIONAL INTERCONNECTION FORUM FOR SERVICES OVER IP (i3 FORUM) Workstream Technical Aspects Workstream Operations Interoperability Test Plan for International Voice services (Release 6) May 2014 Interoperability
More informationThis document specifies the software requirements of CrossTalk+ A VoIP softphone. It describes the specifications of all components of CrossTalk.
1. Introduction CrossTalk+ is a VoIP (Voice over IP) softphone which lets you call anywhere in the world at nominal rates. CrossChat the chat component of CrossTalk enables you to chat with people speaking
More informationVoice over IP. Demonstration 1: VoIP Protocols. Network Environment
Voice over IP Demonstration 1: VoIP Protocols Network Environment We use two Windows workstations from the production network, both with OpenPhone application (figure 1). The OpenH.323 project has developed
More informationCrossTalk is a VoIP (Voice over IP) softphone which lets you call anywhere in the world at nominal rates.
1. Introduction CrossTalk is a VoIP (Voice over IP) softphone which lets you call anywhere in the world at nominal rates. 1.1 Purpose This document specifies the software requirements of CrossTalk v1.04
More informationGoIP Series. SIM Card for GSM Voice Gateway. User Manual
GoIP Series SIM Card for GSM Voice Gateway User Manual Content Content... 1 1 Overview... 4 1.1 Introduction... 4 1.2 Protocols... 5 1.3 Hardware Feature... 5 1.4 Software Feature... 5 1.5 Product Package
More informationNetComm V90 VoIP Phone Quick Start Guide Draft Release 0.1
NetComm V90 VoIP Phone Quick Start Guide Draft Release 0.1 Copyright NetComm Ltd Overview NetComm V90 SIP VoIP Phone User Guide Table of Contents Overview... 3 V90 VoIP Phone Specification...4 Shipping
More informationipecs Communicator Installation and Operation Guide Please read this manual carefully before operating your set. Retain it for future reference.
ipecs Communicator Installation and Operation Guide ipecs is an Ericsson-LG Brand Please read this manual carefully before operating your set. Retain it for future reference. Revision History Issue Date
More informationVOIP-211RS/210RS/220RS/440S. SIP VoIP Router. User s Guide
VOIP-211RS/210RS/220RS/440S SIP VoIP Router User s Guide Trademarks Contents are subject to revise without prior notice. All trademarks belong to their respective owners. FCC Warning This equipment has
More informationinternet technologies and standards
Institute of Telecommunications Warsaw University of Technology 2015 internet technologies and standards Piotr Gajowniczek Andrzej Bąk Michał Jarociński multimedia in the Internet Voice-over-IP multimedia
More informationAn Introduction to VoIP Protocols
An Introduction to VoIP Protocols www.netqos.com Voice over IP (VoIP) offers the vision of a converged network carrying multiple types of traffic (voice, video, and data, to name a few). To carry out this
More informationVoice Over IP. Priscilla Oppenheimer www.priscilla.com
Voice Over IP Priscilla Oppenheimer www.priscilla.com Objectives A technical overview of the devices and protocols that enable Voice over IP (VoIP) Demo Packet8 and Skype Discuss network administrator
More informationRelease Notes for NeoGate TE100 16.18.0.X
Release Notes for NeoGate TE100 16.18.0.X ===Firmware Version: V16.18.0.2==== Applicable Model: NeoGate TE100 Release Date: October 25th, 2014 http://www.yeastar.com 1/6 1. New Features 1. Added support
More informationFunctional Specifications Document
Functional Specifications Document VOIP SOFT PBX Project Code: SPBX Project Advisor : Aftab Alam Project Team: Umair Ashraf 03-1853 (Team Lead) Imran Bashir 02-1658 Khadija Akram 04-0080 Submission Date:19-10-2007
More informationONcbx Feature Guide UC Desktop Client
1 Getting Started 1.1 Installation The Quick Start Guide contains the essential information for getting started with the Oxford Networks BroadTouch Business Communicator. Once you receive an email indicating
More informationKeywords: VoIP calls, packet extraction, packet analysis
Chapter 17 EXTRACTING EVIDENCE RELATED TO VoIP CALLS David Irwin and Jill Slay Abstract The Voice over Internet Protocol (VoIP) is designed for voice communications over IP networks. To use a VoIP service,
More informationWhite paper. SIP An introduction
White paper An introduction Table of contents 1 Introducing 3 2 How does it work? 3 3 Inside a normal call 4 4 DTMF sending commands in sip calls 6 5 Complex environments and higher security 6 6 Summary
More informationTroubleshooting Tools to Diagnose or Report a Problem February 23, 2012
Troubleshooting Tools to Diagnose or Report a Problem February 23, 2012 Proprietary 2012 Media5 Corporation Scope of this Document This Technical Bulletin aims to inform the reader on the troubleshooting
More informationCisco PAP2T Internet Phone Adapter with 2 VoIP Ports Cisco Small Business Voice Gateways and ATAs
Cisco PAP2T Internet Phone Adapter with 2 VoIP Ports Cisco Small Business Voice Gateways and ATAs Feature-Rich VoIP Service Through Your High-Speed Internet Connection Highlights Enables high-quality feature-rich
More informationProject Code: SPBX. Project Advisor : Aftab Alam. Project Team: Umair Ashraf 03-1853 (Team Lead) Imran Bashir 02-1658 Khadija Akram 04-0080
Test Cases Document VOIP SOFT PBX Project Code: SPBX Project Advisor : Aftab Alam Project Team: Umair Ashraf 03-1853 (Team Lead) Imran Bashir 02-1658 Khadija Akram 04-0080 Submission Date:23-11-2007 SPBX
More informationNote: these functions are available if service provider supports them.
Key Feature New Feature Remote Maintenance: phone can be diagnosed and configured by remote. Zero Config: automated provisioning and software upgrading even through firewall/nat. Centralized Management:
More informationVoice over IP. Presentation Outline. Objectives
Voice over IP Professor Richard Harris Presentation Outline Brief overview of VoIP and applications Challenges of VoIP IP Support for Voice Protocols used for VoIP (current views) RTP RTCP RSVP H.323 Semester
More information2100 Series VoIP Phone
2100 Series VoIP Phone Installation and Operations Manual Made in the USA 3 Year Warranty N56 W24720 N. Corporate Circle Sussex, WI 53089 RP8500SIP 800-451-1460 262-246-4828 (fax) Ver. 4 www.rathmicrotech.com
More informationCurso de Telefonía IP para el MTC. Sesión 2 Requerimientos principales. Mg. Antonio Ocampo Zúñiga
Curso de Telefonía IP para el MTC Sesión 2 Requerimientos principales Mg. Antonio Ocampo Zúñiga Factors Affecting Audio Clarity Fidelity: Audio accuracy or quality Echo: Usually due to impedance mismatch
More informationUnit 23. RTP, VoIP. Shyam Parekh
Unit 23 RTP, VoIP Shyam Parekh Contents: Real-time Transport Protocol (RTP) Purpose Protocol Stack RTP Header Real-time Transport Control Protocol (RTCP) Voice over IP (VoIP) Motivation H.323 SIP VoIP
More informationFeatures Reference. About Unified Communication System. Before Using This Machine. Starting a Meeting. What You Can Do During the Meeting
Features Reference About Unified Communication System Before Using This Machine Starting a Meeting What You Can Do During the Meeting Leaving a Meeting Managing Address Book Changing Network Configuration
More informationFollow these steps to prepare the module and evaluation board for testing.
2 Getting Started 2.1. Hardware Installation Procedure Follow these steps to prepare the module and evaluation board for testing. STEP1: Plug the EG-SR-7100A module into the sockets on the test board.
More informationGuide to TCP/IP, Third Edition. Chapter 3: Data Link and Network Layer TCP/IP Protocols
Guide to TCP/IP, Third Edition Chapter 3: Data Link and Network Layer TCP/IP Protocols Objectives Understand the role that data link protocols, such as SLIP and PPP, play for TCP/IP Distinguish among various
More informationSIP SOFTPHONE SDK Apple MAC Desktop OS
SIP SOFTPHONE SDK Apple MAC Desktop OS TECHNICAL DOCUMENTATION VERSION 1.4 November 2014 Page 1 of 69 CONTENTS INTRODUCTION AND QUICK START... 4 EXPORTED FUNCTIONS... 5 InitializeEx()... 5 RegisterToProxy()...
More informationHow to make free phone calls and influence people by the grugq
VoIPhreaking How to make free phone calls and influence people by the grugq Agenda Introduction VoIP Overview Security Conclusion Voice over IP (VoIP) Good News Other News Cheap phone calls Explosive growth
More informationNETWORK ADMINISTRATION
NETWORK ADMINISTRATION INTRODUCTION The PressureMAP software provides users who have access to an Ethernet network supporting TCP/IP with the ability to remotely log into the MAP System via a network connection,
More informationVOIP Project: System Design
VOIP Project: System Design Sarfraz Nawaz Mark Niebur Scott Schuff Athar Shiraz Siddiqui Overview The purpose of this document is to describe in detail the design of our CSEE 4840 semester project: a voice-over-ip
More informationSIP-T22P User s Guide
SIP-T22P User s Guide Thank you for choosing this T-22 Enterprise IP Phone. This phone is especially designed for active users in the office environment. It features fashionable and sleek design, and abundant
More informationCISCO SPA3102 PHONE ADAPTER WITH ROUTER
CISCO SMALL BUSINESS VOICE GATEWAYS AND ATAS Intelligent Call-Routing Gateway for VoIP HIGHLIGHTS Enables high-quality, feature-rich voice-over-ip service through your broadband Internet connection Two
More informationHow to Configure the Allworx 6x, 24x and 48x for use with Integra Telecom SIP Solutions
How to Configure the Allworx 6x, 24x and 48x for use with Integra Telecom SIP Solutions Overview: This document provides a reference for configuration of the Allworx 6x IP PBX to connect to Integra Telecom
More informationWEB CONFIGURATION. Configuring and monitoring your VIP-101T from web browser. PLANET VIP-101T Web Configuration Guide
WEB CONFIGURATION Configuring and monitoring your VIP-101T from web browser The VIP-101T integrates a web-based graphical user interface that can cover most configurations and machine status monitoring.
More informationSIP : Session Initiation Protocol
: Session Initiation Protocol EFORT http://www.efort.com (Session Initiation Protocol) as defined in IETF RFC 3261 is a multimedia signaling protocol used for multimedia session establishment, modification
More informationCisco SPA901 1-Line IP Phone Cisco Small Business IP Phone
Cisco SPA901 1-Line IP Phone Cisco Small Business IP Phone Durable, Affordable, Feature-Rich IP Telephone for the Home Office and Business Small, affordable, single line business class IP Phone Connect
More informationProcedure: You can find the problem sheet on Drive D: of the lab PCs. 1. IP address for this host computer 2. Subnet mask 3. Default gateway address
Objectives University of Jordan Faculty of Engineering & Technology Computer Engineering Department Computer Networks Laboratory 907528 Lab.4 Basic Network Operation and Troubleshooting 1. To become familiar
More informationEthernet. Ethernet. Network Devices
Ethernet Babak Kia Adjunct Professor Boston University College of Engineering ENG SC757 - Advanced Microprocessor Design Ethernet Ethernet is a term used to refer to a diverse set of frame based networking
More informationInternetworking and IP Address
Lecture 8 Internetworking and IP Address Motivation of Internetworking Internet Architecture and Router Internet TCP/IP Reference Model and Protocols IP Addresses - Binary and Dotted Decimal IP Address
More informationVoIP Interkonnektion Test Specification
VoIP Interkonnektion Specification Ausgabedatum 310.2015 Ersetzt Version - Gültig ab 012015 Vertrag Vertrag betreffend Verbindung von VoIP Fernmeldeanlagen und -diensten Gültig ab 012015 1/21 Table of
More informationThis specification this document to get an official version of this User Network Interface Specification
This specification describes the situation of the Proximus network and services. It will be subject to modifications for corrections or when the network or the services will be modified. Please take into
More informationConfiguring SIP Trunking and Networking for the NetVanta 7000 Series
61200796L1-29.4E July 2011 Configuration Guide Configuring for the NetVanta 7000 Series This configuration guide describes the configuration and implementation of Session Initiation Protocol (SIP) trunking
More informationGrandstream Networks, Inc.
Grandstream Networks, Inc. GVC3200/GVC3200 Conferencing System for Android TM Application Note: Preliminary Interoperability Test between GVC3200/GVC3200 and Other Video Conference Systems Index INTRODUCTION...
More informationDeployment Guide for Maximum Security Environments Polycom HDX Systems, Version 3.0.5
Polycom HDX Systems, Version 3.0.5 A warning about operating in a maximum security environment The maximum security profile is designed to lock down communications to the most stringent requirements of
More informationIP-Telephony SIP & MEGACO
IP-Telephony SIP & MEGACO Bernard Hammer Siemens AG, Munich Siemens AG 2001 1 Presentation Outline Session Initiation Protocol Introduction Examples Media Gateway Decomposition Protocol 2 IETF Standard
More informationTOE2-IP FTP Server Demo Reference Design Manual Rev1.0 9-Jan-15
TOE2-IP FTP Server Demo Reference Design Manual Rev1.0 9-Jan-15 1 Introduction File Transfer Protocol (FTP) is the protocol designed for file sharing over internet. By using TCP/IP for lower layer, FTP
More informationCisco SPA942 4-Line IP Phone with 2-Port Switch Cisco Small Business IP Phones
Cisco SPA942 4-Line IP Phone with 2-Port Switch Cisco Small Business IP Phones Advanced, Feature-Rich, Multiline IP Phone for SIP-Based VoIP Service Highlights Industry-leading VoIP technology from Cisco
More informationHardware Features Voicemail message waiting indicator light Voicemail message retrieval button Volume control Redial Button Flash Button Standard
Hardware Features Voicemail message waiting indicator light Voicemail message retrieval button Volume control Redial Button Flash Button Standard 12-button dialing pad High-quality handset One Ethernet
More informationGW400 VoIP Gateway. User s Guide
GW400 VoIP Gateway User s Guide P/N: 956YD30001 Copyright 2006. All Rights Reserved. Document Version: 1.0 All trademarks and trade names are the properties of their respective owners. i Table of Contents
More informationHigh Definition PoE IP Phone
High Definition IP Phone Highlights Key Features Supports SIP 2.0 (RFC3261) Supports 1 SIP voice line IEEE 802.3af Power over Ethernet compliant Supports HD voice LDAP / TR-069 / SNMP Cost-effective, High-performance
More informationPIKA HMP 3.0 High Level API Programmer's Guide
Copyright (c) 2011. All rights reserved. Table of Contents 1 Copyright Information 1 2 Contacting PIKA Technologies 2 3 Introduction 3 3.1 Purpose and Scope 4 3.2 Assumed Knowledge 4 3.3 Related Documentation
More informationESI SIP Trunking Installation Guide
ESI SIP Trunking Installation Guide 0450-1227 Rev. B Copyright 2009 ESI (Estech Systems, Inc.). Information contained herein is subject to change without notice. ESI products are protected by various U.S.
More informationVoIP Ceiling Speaker with Allworx 6x Server Setup Guide
VoIP Ceiling Speaker with Allworx 6x Server Setup Guide CyberData Corporation 2555 Garden Road Monterey, CA 93940 T:831-373-2601 F: 831-373-4193 www.cyberdata.net 2 1.0 Setup Diagram Figure 1-1 is a setup
More informationSIP Trunking. Service Guide. www.megapath.com. Learn More: Call us at 877.634.2728.
Service Guide Learn More: Call us at 877.634.2728. www.megapath.com What is MegaPath SIP Trunking? SIP Trunking enables your business to reduce costs and simplify IT management by combining voice and Internet
More information2100-9 Series VoIP Phone
2100-9 Series VoIP Phone Installation and Operations Manual Made in the USA 3 Year Warranty N56 W24720 N. Corporate Circle Sussex, WI 53089 RP8500SIP 800-451-1460 262-246-4828 (fax) Ver. 3 12/14 Thank
More informationAdvanced, Affordable, Feature Rich IP Phone for the Home Office and Business
IP Phone PRODUCT DATA Advanced, Affordable, Feature Rich IP Phone for the Home Office and Business Comprehensive Interoperability and SIP Based Feature Set Based on the SIP standard, the has been tested
More informationUsing Advanced Phone Features
Using Advanced Phone Features This chapter describes how to configure advanced features on your IP Phone. It contains the following sections: Configuring Privacy and Security, page 9 Enabling and Using
More informationKX-UT113/KX-UT123 KX-UT133/KX-UT136 KX-UT248
Model No. SIP Phone KX-UT113/KX-UT123 KX-UT133/KX-UT136 KX-UT248 Thank you for purchasing this Panasonic product. Please read this manual carefully before using this product and save this manual
More informationVoIP Telephone Adapter User s Manual
VoIP Telephone Adapter User s Manual Last Update: 2008/10/10 1 Introduction...3 1.1 Product Overview (Single Phone Port Model)...3 1.2 Product Overview (Dual Phone Port Model)...4 2 IVR Interface for TA...6
More informationIP - The Internet Protocol
Orientation IP - The Internet Protocol IP (Internet Protocol) is a Network Layer Protocol. IP s current version is Version 4 (IPv4). It is specified in RFC 891. TCP UDP Transport Layer ICMP IP IGMP Network
More informationPolycom SoundPoint IP 550
Polycom SoundPoint IP 550 User Guide For training/documentation, please visit us @ http://customertraining.verizonbusiness.com or call 1 800 662 1049 2009 Verizon. All Rights Reserved. The Verizon and
More informationIP Telephony Deployment Models
CHAPTER 2 Sections in this chapter address the following topics: Single Site, page 2-1 Multisite Implementation with Distributed Call Processing, page 2-3 Design Considerations for Section 508 Conformance,
More informationCisco SPA2012 Phone Adapter with Router
Cisco SPA2102 Phone Adapter with Router Cisco Small Business Voice Gateways and ATAs Feature-Rich Voice Adapter for VoIP Highlights Enables high-quality, feature-rich voice-over-ip service through your
More informationCisco SPA941 4-Line IP Phone with 1-Port Ethernet Cisco Small Business IP Phones
. Data Sheet Cisco SPA941 4-Line IP Phone with 1-Port Ethernet Cisco Small Business IP Phones Advanced, Affordable, Feature-Rich IP Phone for the Home Office and Business Highlights Full-featured four-line
More informationVoIP ATA series (ATA171plus, ATA172plus, ATA-171, ATA-172, ATA-171M, ATA-171P)
ATA Web User Guide VoIP ATA series (ATA171plus, ATA172plus, ATA-171, ATA-172, ATA-171M, ATA-171P) User Guide Released Date : January-2012 Firmware Version : V.300 1. Introduction... 4 2. Hardware Overview...
More informationCisco SPA921 1-Line IP Phone with Display Cisco Small Business IP Phones
Cisco SPA921 1-Line IP Phone with Display Cisco Small Business IP Phones Advanced, Affordable, Feature-Rich IP Phone for the Home Office and Business Highlights Full-featured one-line business-class IP
More informationFirmware version: 1.10 Issue: 7 AUTODIALER GD30.2. Instruction Manual
Firmware version: 1.10 Issue: 7 AUTODIALER GD30.2 Instruction Manual Firmware version: 2.0.1 Issue: 0.6 Version of the GPRS transmitters configurator: 1.3.6.3 Date of issue: 07.03.2012 TABLE OF CONTENTS
More informationImplementing SIP and H.323 Signalling as Web Services
Implementing SIP and H.323 Signalling as Web Services Ge Zhang, Markus Hillenbrand University of Kaiserslautern, Department of Computer Science, Postfach 3049, 67653 Kaiserslautern, Germany {gezhang, hillenbr}@informatik.uni-kl.de
More informationGSM. Quectel Cellular Engine. GSM TCPIP Application Notes GSM_TCPIP_AN_V1.1
GSM Cellular Engine GSM TCPIP Application Notes GSM_TCPIP_AN_V1.1 Document Title GSM TCPIP Application Notes Version 1.1 Date 2011-09-22 Status Document Control ID Release GSM_TCPIP_AN_V1.1 General Notes
More informationLync for Mac 2011 Deployment Guide
2011 Deployment Guide Getting Started Information in this document, including URL and other Internet Web site references, is subject to change without notice. Content in this document represents the current
More informationCreating your own service profile for SJphone
SJ Labs, Inc. 2005 All rights reserved SJphone is a registered trademark. No part of this document may be copied, altered, or transferred to, any other media without written, explicit consent from SJ Labs
More informationSIP Trunking using Optimum Business Sip Trunk Adaptor and the Zultys MX250 IP PBX
SIP Trunking using Optimum Business Sip Trunk Adaptor and the Zultys MX250 IP PBX Table of Contents Goal 3 Prerequisites 3 Zultys MX250 Configuration 4 Network Settings 4 Phone Registration and Assignment
More informationVoice over IP Probe! for Network Operators and! Internet Service Providers
Voice over IP Probe! for Network Operators and! Internet Service Providers Product Presentation September 2011 2011 ADVENAGE GmbH Agenda Voice over IP Probe Key Facts VoIP Probe in a Nutshell Use Cases
More informationPacket Capture. Document Scope. SonicOS Enhanced Packet Capture
Packet Capture Document Scope This solutions document describes how to configure and use the packet capture feature in SonicOS Enhanced. This document contains the following sections: Feature Overview
More informationIP Phone. Installer Guide. LIP-8002/8002A SIP (Session Initiation Protocol) ipecs
IP Phone SIP (Session Initiation Protocol) ipecs Please read this manual carefully before installation. Retain it for future reference. Regulatory and Safety Notices 1. Radio Frequency Emissions FCC Compliance
More informationAvaya IP Office SIP Trunk Configuration Guide
Valcom Session Initiation Protocol (SIP) VIP devices are compatible with SIP-enabled versions of Avaya IP Office (5.0 and higher). The Valcom device can be added to the IP Office system as a SIP Trunk.
More informationHigh Definition PoE IP Phone
High Definition IP Phone Key Features Highlights Supports SIP 2.0 (RFC3261) IEEE 802.3af Power over Ethernet compliant (VIP- 1000PT only) Supports HD voice (G.722) Voice Activity Detection Auto Provisioning:
More informationVoIP with SIP. Session Initiation Protocol RFC-3261/RFC-2543. Tasuka@Tailyn.com.tw
VoIP with SIP Session Initiation Protocol RFC-3261/RFC-2543 Tasuka@Tailyn.com.tw 1 Legacy Telephone 2 Legacy Telephone 2 Legacy Telephone 2 Legacy Telephone 2 Legacy Telephone 2 Legacy Telephone 2 Legacy
More informationSHORT DESCRIPTION OF THE PROJECT...3 INTRODUCTION...4 MOTIVATION...4 Session Initiation Protocol (SIP)...5 Java Media Framework (JMF)...
VoIP Conference Server Evgeny Erlihman jenia.erlihman@gmail.com Roman Nassimov roman.nass@gmail.com Supervisor Edward Bortnikov ebortnik@tx.technion.ac.il Software Systems Lab Department of Electrical
More informationTo ensure you successfully install Timico VoIP for Business you must follow the steps in sequence:
To ensure you successfully install Timico VoIP for Business you must follow the steps in sequence: Firewall Settings - you may need to check with your technical department Step 1 Install Hardware Step
More informationMITEL SIP CoE Technical. Configuration Note. Configure MCD for use with Thinktel SIP Trunking Service. SIP CoE 12-4940-00197
MITEL SIP CoE Technical Configuration Note Configure MCD for use with SIP Trunking Service SIP CoE NOTICE The information contained in this document is believed to be accurate in all respects but is not
More informationACP ThinManager Tech Notes Troubleshooting Guide
ACP ThinManager Tech Notes Troubleshooting Guide Use the F1 button on any page of a ThinManager wizard to launch Help for that page. Visit www.thinmanager.com/technotes/ to download the manual, manual
More information3 Specification. Master Chip Broadcom 1190. Keys Direct-button 1. Microphone 1. Amplifier 0.5W Speaker 0.5W. Voice
I20-T Door Phone 1 Description I20-T Voice Access control is a digital IP door phone with Fanvil Voip Solutions (Broadcom chip 1190), stable and reliable. FDSP ensure voice loud and clear. Good looking,
More informationInter-Tel. 3000 IP Phone Quick Reference Guide
Inter-Tel 3000 IP Phone Quick Reference Guide Introduction to your IP Phone The Inter-Tel 3000 IP Phone provides the same functionality as the Inter-Tel 3000 digital telephone sets. It can be connected
More informationVoIP Handset User Manual. Digital Voice Quality Business Grade Handset Easy Plug & Play
Digital Voice Quality Business Grade Handset Easy Plug & Play Table of Contents 1.0 Getting Started... 1 1.1 Unpacking the box... 1 1.2 Assembling the Phone... 2 2.0 Phone Button Features... 3 3.0 Phone
More information