IP telephony

H.323 Protocol

 

Telesis Systems Offering H.323 Protocol

• Telesis Business Phone Systems:
  • PX24N (Telesis Nano) IP PBX SoHo System
  • PX24U Hybrid IP PBX Business Phone System
  • PX24M Hybrid IP PBX Business Phone System
  • PX24X Hybrid IP PBX Business Phone System
• Telesis Switching Systems:
  • X1 Large Capacity TDM - IP Telephony Switch
• Combined VoIP Gateway and Signaling Converters:
  • Stillink 200
  • Stillink 800
  • Stillink 3200

Standards

The applied standards for H.323 umbrella protocol are H.225.0 (07/2003) version 5, H.235 (05/2003) version 3, H.245 (07/2003) version 12, H.450 (09/1997), AES FIPS PUB 197.

Applications

Telesis systems integrate both packet and circuit switching technology. A Telesis system featuring an integrated gatekeeper provides an economical way for administrators to manage a central database of phone numbers without the expense of a separate-box gatekeeper solution. Number/IP translation is performed through Telesis advanced routing algorithm. Together with the integrated gatekeeper, call authorization, call management, enhanced billing functions, flexible routing algorithms and extensive business telephony features make a Telesis system serve as a feature-rich communication platform. Integrated gatekeeper allows calls to be placed direct or gatekeeper routed between H.323 entities in various circumstances.

The integrated H.323 gatekeeper may serve to numerous H.323 users

Telesis systems can register to multiple H.323 gatekeepers at the same time. This allows address resolution of a Telesis system from either side and results in flexibility for multipath VoIP access applications.

Capability of registering to multiple H.323 gatekeepers at the same time provides multipath VoIP access

 

 Telesis systems support numerous H.323 users (entities) which can be terminals, gatekeepers or gateways.

H.323 entities may be terminals, gatekeepers, gateways at the same time

Furthermore, Telesis systems are with media proxying capabilities. The integrated media proxy provides a transit point for media (audio) streams between H.323 entities. The media proxy operates only for the integrated gatekeeper routed calls in some circumstances.

While voice bridging distant offices over the IP, security of a VoIP call is guaranteed with the encryption (optional) of voice according to 256 bit AES (AES-256).

Telesis systems support AES 256 media encryption over H.323

While connecting to the long distance call operator over the IP, the Telesis system may register to an external gatekeeper of the operator as an option. With the advanced routing algorithms and alternate routing capability, TDM calls from a terminal equipment of the system may be routed to a selected operator over the IP or PSTN. Alternate routing capability provides automatic fall back to the PSTN if the IP network is unaccessible.

Advanced routing algorithms and alternate routing capability ensure call progress

Layers and Options

The physical layer of H.323 protocol in Telesis systems is 10/100 BaseT Ethernet. Several ethernet and other properties for H.323 are programmable.

Audio Codecs

A Telesis system is equipped with well-known audio codecs featuring audio compression as well. Audio codec preference list and properties such as silence suppression (VAD-Voice Activity Detection), frame length are programmable for the system. Currently available codecs for VoIP calls are:

• G.711 (A and u)
• G.723.1 (5.3kbps, 6.4kbps)
• G.729
• G.729AB

Echo Cancellation

An AT&T certified G.168 echo canceler meets and exceeds G.168-2002 standards. The echo canceler can operate with delays as high as 128msec. It is better than industry standard cancelers under the most important and difficult conditions like double-talk and the presence of background noise.

Route and Routeset Configuration

In a Telesis system, an H.323 endpoint may have its own route number. It is possible to define numerous distinct routes. A given route to a particular destination and its accompanying alternate routes are grouped in a routeset. Each route in a routeset has a priority order. Alternate routes may be H.323 endpoints or TDM (PSTN) lines. Routing to the next priority alternate route is possible in the event that a route becomes unavailable.

Call Routing

As an IP-TDM gateway, the Telesis system routes a call from the TDM network to the IP network according to:

• Dialed digits (called number)
• DPC

• Calling party number and information elements whenever available, such that:

  • Category of calling party
  • NOA of calling party
  • Numbering plan of calling party
  • Presentation status of calling party
  • Screening status of calling party

H.450 Supplementary Services

Telesis systems support various H.450.x series supplementary services in H.323 protocol stack, such as:

• H.450.1 Generic functional protocol for the support of supplementary services in H.323
• H.450.2 Call transfer supplementary service for H.323
• H.450.4 Call hold supplementary service for H.323

 

 

SIP - Session Initiation Protocol

 

Telesis Systems Offering SIP

• Telesis Business Phone Systems:
  • PX24N (Telesis Nano) IP PBX SoHo System
  • PX24U Hybrid IP PBX Business Phone System
  • PX24M Hybrid IP PBX Business Phone System
  • PX24X Hybrid IP PBX Business Phone System
• Telesis Switching Systems:
  • X1 Large Capacity TDM - IP Telephony Switch
• Combined VoIP Gateway and Signaling Converters:
  • Stillink 200
  • Stillink 800
  • Stillink 3200

Standards

RFC 3261 SIP: Session Initiation Protocol.

Applications

Telesis systems integrate both packet and circuit switching technology. A Telesis system featuring an integrated SIP registrar/proxy/server provides an economical way for administrators to manage a central database of phone numbers without the expense of a separate-box registrar/proxy/server solution.

The integrated SIP registrar/proxy/server may serve to numerous SIP user agents

Telesis systems can register to multiple SIP registrars/proxies/servers at the same time. This allows address resolution of a Telesis system from either side and results in flexibility for multipath VoIP access applications.

Capability of registering to multiple SIP registrars/proxies/servers at the same time provides multipath VoIP access

Telesis systems support numerous SIP users (entities) which can be user agents or registrars/proxies/servers. Number/IP translation is performed through an advanced routing algorithm. Together with the integrated registrar/proxy/server, call authorization, call management, enhanced billing functions, flexible routing algorithms, and extensive business telephony features make a Telesis system serve as a feature-rich communication platform.

SIP entities may be user agents and registrars/proxies/servers at the same time

Connecting to the long distance call operator can be both over TDM and IP at the same time. The Telesis system may register at the external registrar/proxy/server of the operator as an option. With the advanced routing algorithms and alternate routing capability, TDM calls from a terminal equipment connected to the system may be routed to a selected operator over the IP or PSTN. Alternate routing capability provides automatic fall back to the PSTN if the IP network is unaccessible.

Advanced routing algorithms and alternate routing capability ensure call progress 

Another SIP related feature powering Telesis systems is the embedded WepPhone service. Telesis WebPhone is a SIP softphone or client applet hosted in Telesis IP PBX, Telesis TDM - IP telephony systems, Stillink access gateways and Stillink signaling converters. The applet can be run from any Java enabled web browser. Since it is a standard Java applet, no software installation is required or no plugin is installed. The applet is just loaded temporarily onto the client`s browser. It can be used wherever is an access to the Internet. A web browser, a microphone, and a headset are sufficient to make calls.

Integrated Web Server of the Telesis System Hosts the Java SIP Client Applet

Layers and Options

The physical layer of SIP protocol in Telesis systems is 10/100 BaseT Ethernet. Several ethernet and other properties for SIP are programmable.

Audio Codecs

Telesis systems are equipped with well-known audio codecs featuring audio compression as well. Audio codec preference list and properties such as silence suppression (VAD-Voice Activity Detection), frame length are programmable for the system. Currently available codecs for VoIP calls are:

• G.711 (A and u)
• G.723.1 (5.3kbps, 6.4kbps)
• G.729
• G.729AB

Echo Cancellation

An AT&T certified G.168 echo canceler meets and exceeds G.168-2002 standards. The echo canceler can operate with delays as high as 128msec. It is better than industry standard cancelers under the most important and difficult conditions like double-talk and the presence of background noise.

Route and Routeset Configuration

In a Telesis system, a SIP user agent may have its own route number. It is possible to define numerous distinct routes. A given route to a particular destination and its accompanying alternate routes are grouped in a routeset. Each route in a routeset has a priority order. Alternate routes may be SIP user agents or TDM (PSTN) lines. Routing to the next priority alternate route is possible in the event that a route becomes unavailable.

Call Routing

A Telesis system routes a call from the TDM network to the IP network according to:

• Dialed digits (called number)
• DPC
• Calling party number and information elements whenever available, such that:
  • Category of calling party
  • NOA of calling party
  • Numbering plan of calling party
  • Presentation status of calling party
  • Screening status of calling party

 

H.323 and SIP Integration

 

Telesis Systems with H.323 - SIP Integration Capability

• Telesis Business Phone Systems:
  • PX24N (Telesis Nano) IP PBX SoHo System
  • PX24U Hybrid IP PBX Business Phone System 
  • PX24M Hybrid IP PBX Business Phone System 
  • PX24X Hybrid IP PBX Business Phone System
• Telesis Switching Systems:
  • X1 Large Capacity TDM - IP Telephony Switch
• Combined VoIP Gateway and Signaling Converters:
  • Stillink 200
  • Stillink 800
  • Stillink 3200

Description

Telesis systems support for SIP and H.323 protocols. Both protocols coexist on the same Telesis system. SIP and H.323 calls may originate and terminate in the same system. Furthermore, Telesis systems allow calls from SIP based devices to be routed to H.323 based devices and vice versa. With this interoperability, enterprises may have the ability to use both protocols in the same network. 

General Capabilities

Telesis systems can register to both SIP registrar/proxy/servers and H.323 gatekeepers at the same time. This allows address resolution of a Telesis system from either side and results in flexibility for multipath VoIP access applications. Furthermore, Telesis systems have both integrated H.323 gatekeeper and external gatekeeper registration capability at the same time. Similarly, Telesis systems have both integrated SIP registrar and external registrar registration capability at the same time. Coexistence of all these capabilities allows:

• SIP users to call SIP users in private address space
• SIP users to call H.323 users in private address space
• SIP users to call SIP entities in public network
• SIP users to call H.323 entities in public network
• SIP users to call legacy PSTN via TDM interfaces 
• H.323 users to call H.323 users in private address space
• H.323 users to call SIP users in private address space
• H.323 users to call H.323 entities in public network
• H.323 users to call H.323 entities in public network
• H.323 users to call legacy PSTN via TDM interfaces
• Legacy PSTN users to call any H.323 and SIP entities

A Telesis system has both integrated SIP registrar and H.323 gatekeeper for users. Furthermore, it is possible to register to multiple SIP registrars and H.323 gatekeepers at the same time.

Telesis systems try to have the media transport directly between the connecting IP entities. However, in some cases like IP-TDM, TDM-IP, SIP-H.323, H.323-SIP calls, this is not possible or efficient. In such cases, switching and media proxying capabilities in Telesis systems route calls from one side to the other.

 

Xymphony - API

 

Telesis Systems Offering API Server

• Telesis Business Phone Systems:
  • PX24N (Telesis Nano) IP PBX SoHo System
  • PX24U Hybrid IP PBX Business Phone System 
  • PX24M Hybrid IP PBX Business Phone System
  • PX24X Hybrid IP PBX Business Phone System
• Telesis Switching Systems:
  • X1 Large Capacity TDM - IP Telephony Switch

 

Xymphony API in Brief

Telesis proprietary Xymphony-API server in Telesis systems performs various telephony tasks such as dialing, call transfer, call park, call retrieve, conference, conversation recording, and display of port status. The physical connection is between a user`s personal computer, the same user`s analog/digital telephone line and the Telesis system. Both the Xymphony-API server and the PC with Xymphony-API client utility (software) must be in the same local area network. There can be multiple users or client PCs served by the same Xymphony-API server. XCom (Xymphony Companion), which is a Xymphony-API client software developed by Telesis, is an advanced screen-based telephone dialer and console with numerous features to integrate the computer and Telesis system for telephony needs. Moreover, the XCom is a freeware utility.

CRM Integration

Telesis XCom, which is a freeware API client for Telesis systems, is capable of searching automatically SugarCRM records in the common database according to the received Caller ID information. SugarCRM is the world`s leading commercial-open source Customer Relations Management software. SugarCRM is available as a free download as well as a commercial package. The integrated solution is made up of three main components:

• A Telesis system with the Xymphony-API server
• Another Server PC running SugarCRM,
• SugarCRM and XCom users with individual PCs

 

AES 256 Media Encryption

 

Telesis Systems Offering AES 256 Media Encryption

• Telesis Business Phone Systems:
  • PX24N (Telesis Nano) IP PBX SoHo System
  • PX24U Hybrid IP PBX Business Phone System
  • PX24M Hybrid IP PBX Business Phone System
  • PX24X Hybrid IP PBX Business Phone System
• Telesis Switching Systems:
  • X1 Large Capacity TDM - IP Telephony Switch
• Combined VoIP Gateway and Signaling Converters:
  • Stillink 200
  • Stillink 800
  • Stillink 3200
• IP Phones
  • Telesis ITS821 Executive IP Telephone Set
  • XPhone Softphone PC Edition
  • XPhone Softphone Pocket PC Edition
  • XPhone Softphone Smartphone Edition    

VoIP Protocol inTelesis Systems Offering AES 256 Media Encryption

• H.323
• xSIP (eXtended SIP)

H.323 and AES 256 Media Encryption

Introduction

All Telesis systems are complete voice communication systems, which combine various TDM interfaces and IP components. They are all-in-one solutions with integrated gatekeeper, softswitch capability, IP-TDM routing (gateway) functions, and numerous IP and traditional system features. Even though the media encrypting algorithm explained here is applicable for H.323 endpoint-to-endpoint connection too, it is recommended for H.323 endpoint-to-gatekeeper connection for further security.

The following paragraphs demonstrate algorithms applied for site-to-site communication in brief, such that:

• Two Telesis systems in each site
• Both systems are provided with necessary licenses for the VoIP media security and their parameters are set accordingly.

While voice bridging distant offices over the IP, security of a VoIP call is guaranteed with the encryption of voice according to 256 bit AES (AES-256).

Telesis systems support AES 256 media encryption over H.323

Secure Gatekeeper Registration

Two Telesis systems share an account name and a secret, which is the password. One system as an H.323 endpoint registers to the gatekeeper of the other with the shared account name and the password. For the registration, H.225 RAS messages are exchanged between the two Telesis systems according to the H.235 Baseline Security Profile with or without integrity check. The baseline security profile provides basic security for endpoint-to-gatekeeper registration using the secure password-based HMAC-SHA1-96 hashing algorithm.

Baseline authentication

For H.323 endpoint-to-gatekeeper registration, RAS message authentication is according to H.235 Baseline Security Profile standards. This security service supports authentication of selected fields only, but does not provide full message integrity. The authentication-only security profile may be preferable for the messages traversing NAT/firewall devices. Hashing algorithm is the password-based HMAC-SHA1-96.

Baseline integrity

For H.323 endpoint-to-gatekeeper registration, RAS message authentication and integrity is according to H.235 Baseline Security Profile standards. This is a security combining both message integrity and the authentication. Hashing algorithm is the password-based HMAC-SHA1-96.

Encrypting the Media

For encrypting the media, 256-bit Advanced Encryption Standard (AES-256) is used. AES-256 specifies a cryptographic algorithm using a symmetrical block cipher that can process data blocks of 128 bits with 256bit chipher (crypto) key which is agreed by Diffie-Hellman procedure. Audio samples are collected from the codec, they are encrypted, and inserted into the RTP payloads. When the receiving side gets RTP payloads, the decrypting occurs.

A secure contact would be by generating and exchanging shared Diffie-Hellman half-keys. Diffie-Hellman master key for the AES-256 encryption is generated from the combination of the two shared half keys exchanged by two Telesis systems involved in a call.

Diffie-Hellman key exchange

Telesis systems exchange Diffie-Hellman half keys using authentication based on H.235 Baseline Security Profile with or without integrity check. This prevents Man-in-the-Middle (MIM) attacks and communicating systems can be sure with whom they share the Diffie-Hellman half keys. Hash algorithm for H.235 Baseline Security Profile or H.235 Baseline Security Profile with integrity check is HMAC-SHA1-96. Exchange of HMAC-SHA1-96 hashed Diffie-Hellman half keys provides additional security.

Key exchange occurs during H323 call signaling (H.225) messaging between two systems for end-to-end communication. First call signaling message in both direction are used in key exchange. Setup message is used in forward direction. Setup Acknowledge, Call proceeding, Alerting or Connect message can be used in reverse direction. Since, the authentication keyed by the password, which is a secret in two systems, it may be open to MIM attacks if simple passwords are chosen. Telesis systems allow Diffie-Hellman half key exchange provided that a sufficiently long password is selected. In the following cases, the call fails before connect.

• Authentication failure
• Authentication but missing half key in Setup message
• Authentication but missing half key in one of Setup Acknowledge, Call proceeding, Alerting or Connect messages

Summary

Security of VoIP communication between two Telesis systems is ensured with:

• A sufficiently long password
• Baseline Security Profile for RAS messaging for H.323 endpoint-to-gatekeeper registration
• Baseline Security Profile for Call Signaling for secure Diffie-Hellman key exchange. 
• Exchange of HMAC-SHA1-96 hashed Diffie-Hellman half keys
• Cipher AES-256

xSIP and AES 256 Media Encryption

xSIP (eXtended SIP) protocol has been developed by Telesis. The main purpose of its development is to make some value-added services in Telesis systems to be applicable for VoIP calls too.

Telesis ITS821 IP telephones connected to a Telesis system. AES 256 media encryption over xSIP

Beyond the comfort and availability of value-added services, xSIP also allows secure communication with utilizing AES-256 media encryption. Telesis Business Phone Systems, TDM-IP Telephony Switches, as well as ITS821 Executive IP Telephone Sets (or XPhone Softphone) support AES-256 over xSIP protocol. While voice bridging distant offices over the xSIP, security of a VoIP call is provided by:

• A Telesis system (where ITS821 IP Telephones or XPhone Softphones register) with the necessary encryption license
• Appropriate firmware (free) installed in the Telesis System
• Appropriate firmware (free) installed in ITS821 IP Telephones
• Appropriate version of XPhone Softphones (PC, Pocket PC, or Smartphone Edition) 

XPhone softphones connected to a Telesis system. AES 256 media encryption over xSIP

Security of VoIP communication between an ITS821 IP Telephone Set (or XPhone Softphone) and a Telesis IP Telephony System is ensured with:

• Telesis developed protocol: xSIP
• Proprietary VoIP codecs
• Intelligent algorithms for authentication
• Exchange of Diffie-Hellman half keys
• Cipher AES-256

The perfect combination of various protocols and algorithms protect your conversations

 

 

xSIP - eXtended SIP

 

Telesis Systems Offering eXtended SIP

• Telesis Business Phone Systems:
  • PX24N (Telesis Nano) IP PBX SoHo System
  • PX24U Hybrid IP PBX Business Phone System 
  • PX24M Hybrid IP PBX Business Phone System
  • PX24X Hybrid IP PBX Business Phone System
• Telesis Switching Systems:
  • X1 Large Capacity TDM - IP Telephony Switch
• Combined VoIP Gateway and Signaling Converters:
  • Stillink 200
  • Stillink 800
  • Stillink 3200

Telesis Phones with eXtended SIP

• Telesis ITS821 Executive IP Telephone Set
• Telesis XPhone VoIP Softphone for PC
• Telesis XPhone Mobile VoIP Softphone for Pocket PC Phone
• Telesis XPhone Mobile VoIP Softphone for Smartphone

Development

xSIP (eXtended SIP) protocol has been developed by Telesis. The main purpose of its development is to make some value-added services in Telesis systems to be applicable for VoIP calls too. Presently, Telesis offers two products with xSIP protocol. One is the ITS821 IP Phone, and the other is the XPhone IP Softphone. Both products bring the comfort of Telesis digital sets over IP. Almost all the functions of Telesis digital telephones are also applicable over IP. To illustrate, ITS821 and XPhone are with a busy display panel showing the current status (idle, busy, ringing, etc.) of 120 pre-programmed users in a Telesis Hybrid IP PBX.

General Services with the xSIP

Telesis systems support numerous xSIP users. Number/IP translation is performed through an advanced routing algorithm. Together with the integrated xSIP registrar, call authorization, call management, enhanced billing functions, flexible routing algorithms, and extensive business telephony features make a Telesis system serve as a feature-rich communication platform.

Extended Services with the xSIP

Some of the extended services with the xSIP protocol and xSIP capable phones are: 

• accessing to missed calls list, dialed numbers list, incoming calls list, and directory stored in the registered Telesis system,
• large LCD display with lots of information about the user and registered Telesis system 
• playing, deleting voice messages and recorded conversations stored in the registered Telesis system, 
• selecting the ringer melody available in the registered Telesis system,
• many programmable function keys.
• many programmable quick access keys and busy display panel (BDP), 
• setting call forward unconditional,
• setting call forward busy,
• setting call forward no-reply,
• setting hotline,
• activating call waiting,
• activating do not disturb,
• activating wake-up (reminder) service,
• observing firmware version of the registered Telesis system,
• deflecting calls,
• activating call back,
• holding calls on,
• retrieving calls,
• transferring calls,
• tracing transferred calls (up to 4 calls),  
• activating conference,
• recording the conversation bi-directionally (both calling and called party voices).

 

ITS821 IP Phone and XPhone IP Softphone (PC edition) provide extended services over IP

XPhone IP Softphone (Mobile edition) for Pocket PC Phone and Smartphone

Layers and Options

The physical layer of xSIP protocol in Telesis systems is 10/100 BaseT Ethernet.

Call Routing

A Telesis system routes a call from the TDM network to the xSIP user according to:

• Dialed digits (called number)
• DPC
• Calling party number and information elements whenever available, such that:
  • Category of calling party
  • NOA of calling party
  • Numbering plan of calling party
  • Presentation status of calling party
  • Screening status of calling party

xSIP and AES 256 Media Encryption

Beyond the comfort and availability of value-added services, xSIP also allows secure communication with utilizing AES-256 media encryption. Telesis Business Phone Systems, TDM-IP Telephony Switches, as well as ITS821 Executive IP Telephone Sets (or XPhone VoIP softphones PC edition) support AES-256 over xSIP protocol.

Telesis ITS821 IP telephones connected to a Telesis system. AES 256 media encryption over xSIP

XPhone softphones connected to a Telesis system. AES 256 media encryption over xSIP

Security of VoIP communication between an ITS821 IP Telephone (or XPhone VoIP softphone PC edition) and a Telesis IP Telephony System is ensured with:

• Proprietary protocol: xSIP
• Proprietary VoIP codes
• Intelligent algorithms for authentication
• Exchange of Diffie-Hellman half keys
• Cipher AES-256

 

 

Automatic Conversation Recording on H.323 and SIP

 

Telesis Systems Offering Conversation Recording on H.323 and SIP

• Telesis Business Phone Systems:
  • PX24N (Telesis Nano) IP PBX SoHo System
  • PX24U Hybrid IP PBX Business Phone System 
  • PX24M Hybrid IP PBX Business Phone System 
  • PX24X Hybrid IP PBX Business Phone System
• Telesis Switching Systems:
  • X1 Large Capacity TDM - IP Telephony Switch
• Combined VoIP Gateway and Signaling Converters:
  • Stillink 200
  • Stillink 800
  • Stillink 3200

Description

Telesis systems allow Automatic Conversation Recording for VoIP calls with much of ease. All VoIP conversations on pre-selected H.323 or SIP entities may be automatically recorded. The solution is:

• fully integrated with the Telesis System
• cost-effective
• easily maintained
• capable of bi-directional, echo-free, high-quality voice recording
• capable of archiving compressed or uncompressed voice records 
• with freeware downloading, archiving, indexing, and reporting utility XTools
• with the possibility of uploading records via FTP
• with the possibility of authorized remote access over IP
• with the possibility of re-selecting H.323 or SIP entities (to be recorded) easily with programming only

Applications

Automatic voice recording of all VoIP calls (conversations) on H.323 or SIP entities may be critical to the organizations, which have migrated or wishing to migrate to IP Telephony. Some operations and applications, where VoIP conversation recording is critical, are;

• Public Safety and Health
• Call Centers
• Air, Maritime, Railway Traffic Control

The integrated DVR (digital voice recorder) within Telesis systems may be used in such operations and applications for the bi-directional (both calling and called party voices) recording of VoIP calls.

Archiving, Indexing, and Reporting Conversation Records

The combination of the integrated DVR hardware (with a storage capacity of 100 hours or more) and an external archiving device is the heart of the solution. In this solution, the integrated DVR operates as the recording buffer. To transfer conversation records to the archiving device, two protocols are availabe:

XTools Protocol

Telesis system acts as a server and listens requests on a predefined UDP port. The client is a PC running the freeware XTools Utility. The XTools Utility collects and archives the conversation records. Indexing and reporting functions may also be handled by the XTools Utility.  

Freeware XTools Utility may collect conversation records from the Telesis system

FTP Protocol

Telesis system acts as a client and periodically tries to connect to an FTP server to transfer recordings in compressed or uncompressed format. In case of compression (for Telesis systems licensed with VoIP codecs), the codec to be used for decompression is VivoActive G.723.1. Otherwise, i.e., uncompressed transmission, the A-Law codec is used. The archiving device can be any FTP server on any operating system. Each conversation record, which is uploaded to the archiving FTP server, has a clear file name. The file name indicates:

• the access code of the user, on which the conversation is recorded
• starting time of recording as year, month, day, hours, minutes, seconds
• a mark if the call is an originated one 
• a mark if the call is a terminated one
• the account id (if available)
• the access code of B party
• numbers dialed by the originating user
• the caller id from originating user

Consequently, software enthusiasts (from novice to profesional) may easily develop their own applications, which may index, filter, report uploaded conversation records.

The embedded FTP client within the Telesis system may upload conversation records to an FTP server

Security Notes

Make sure that the recordings are secure. Keep the PC running the XTools Utility or FTP server for archiving records in a safe place and accessible only by authorized personnel. Automatic conversation (voice) recording function in Telesis systems is technologically secured to prevent its unlicensed and unauthorized use.

 

 

Xymphony - IP DECT Integration

 

Telesis Systems Offering Xymphony - IP DECT Base Station Integration

• Telesis Business Phone Systems:
  • PX24N (Telesis Nano) IP PBX SoHo System
  • PX24U Hybrid IP PBX Business Phone System
  • PX24M Hybrid IP PBX Business Phone System
  • PX24X Hybrid IP PBX Business Phone System
• Telesis Switching Systems:
  • X1 Large Capacity TDM - IP Telephony Switch

Standards

RFC 3261 SIP: Session Initiation Protocol.

Advantages

A Telesis system featuring an integrated SIP registrar/proxy/server together with IP DECT base stations with SIP support provides an economical way for small, medium , and large businesses for wireless users without the expense of a separate-box DECT controller/server solution.

Furthermore, the IP DECT solution:

• eliminates the need for non-standard cabling, a LAN is sufficient to run the base stations,
• simplifies administration and maintenance,
• reduces costs and improves employee mobility and remote connectivity,
• allows remote locations for the mobility.

Telesis PX24 Hybrid IP PBX systems support hundreds of SIP user agents. Similarly, the X1 supports thousands of SIP user agents. These are standard features in Telesis systems and no payment required like VoIP user license fee for each SIP user. Thanks to the support for numerous SIP users as standard, either small or medium or large business may get the benefits of mobile employees without unexpected costs.

Mobile users may also use Telesis features,which are available for wired users. Calling number (Caller ID) may be shown on the mobile DECT handsets, they may hold and transfer calls, leave voice mails or listen voice mails left in their mailboxes.

IP DECT Solutions for All Businesses

Telesis A.S. provides PX24 Hybrid IP PBX Systems and X1 TDM - IP Telephony switches for all size business. Besides, IP DECT solutions are available for small to large business. The reult is a total wireless solution for every organization needing mobility for its employees.

Small Business in a Single Location or in a Single Floor

PX24U or PX24M together with a single cell IP DECT base station type 300 provides a solution for small business with a need to supply up to 12 mobile employees with DECT wireless handsets. The solution is very simple and cost-effective to get the business wireless.

PX24U small capacity Hybrid IP PBX with 16 wired TDM and 12 wireless DECT users. Easy and cost-effective solution for small business.

Small and Medium Businesses in Multiple Floors

PX24U or PX24M together with a single cell IP DECT base station type 300, as well as wireless repeaters, provides a solution up to 12 mobile employees in the multiple floors. Wireless repeaters extend the radio coverage.

A solution for multiple floors with using wireless repeaters. Again, an easy and cost-effective solution for small and medium business.

Medium Business in a Single or Multiple Floors

PX24M or PX24X together with a single cell IP DECT base station type 600 provides a solution for medium business with a need to supply up to 35 mobile employees with DECT wireless handsets. Furthermore, wireless repeaters extend the radio coverage for multiple floors.

More wireless DECT users with using the PX24M medium capacity Hybrid IP PBX and a single IP DECT base station. More radio coverage is provided with wireless repeaters.

Medium and Large Businesses

The PX24X together with multiple cell IP DECT base stations type 600 provides a solution for medium business with a need to supply up to 512 mobile employees with DECT wireless handsets. Furthermore, IP makes it an ideal solution for the business having several locations. Up to 256 radio units can be located to obtain the necessary radio coverage.

Huge number of wireless DECT users with the PX24X large capacity Hybrid IP PBX and multiple IP DECT base stations.

Large Business

The X1 together with multiple cell IP DECT base stations type 600 provides a solution for large business with a need to supply up to 1500 mobile employees with DECT wireless handsets. Furthermore, IP makes it an ideal solution for businesses with several locations. Up to 256 radio units can be located to obtain the necessary radio coverage.

 

 

Telesis WebPhone - SIP Java Client Applet

 

Hallmarks for Clients

• Java based client applet (i.e., softphone) loaded onto the browser
• Works with any Java enabled browser
• No software or plug-in installation onto the client computer
• Just a computer with internet connection, a microphone, and a headset
• No need to worry about updates
• Simple click to call feature, good especially for non-technical people
• DWDI (Direct Web Dialing In) service to call analog, digital, or IP users of a Telesis system

Hallmarks for Servers or Telesis System Owners

• Industry standard SIP (session initiation protocol) technology
• Applet hosted in the Telesis IP Telephony system
• Optimization in Telesis systems to provide best audio quality for calls coming from clients
• Empowering web sites with allowing clients just to click to make calls
• Customization in Telesis system with different click to call templates
• Easy and fast deployment on personal or commercial web sites with simple link to the hosting Telesis system
• Preparing e-mail signatures inviting people to make web calls

Telesis Systems Offering SIP WebPhone Service

• Telesis Business Phone Systems:
  • PX24N (Telesis Nano) IP PBX SoHo System
  • PX24U Hybrid IP PBX Business Phone System
  • PX24M Hybrid IP PBX Business Phone System
  • PX24X Hybrid IP PBX Business Phone System
• Telesis Switching Systems:
  • X1 Large Capacity TDM - IP Telephony Switch
• Combined VoIP Gateway and Signaling Converters:
  • Stillink 200
  • Stillink 800
  • Stillink 3200

Integrated Web Server of the Telesis System Hosts the Java SIP Client Applet

Compatibility

Telesis WebPhone is compatible with all major web browsers such as Google Chrome, Internet Explorer, Mozilla Firefox, Opera, or Safari in Windows, Mac OS, and Linux operating systems.

    

Standards

RFC 3261 SIP: Session Initiation Protocol.

Introduction

Telesis WebPhone is a SIP softphone or client applet hosted in Telesis IP PBX, Telesis TDM - IP telephony systems, Stillink access gateways and Stillink signaling converters. The applet can be run from any Java enabled web browser. Since it is a standard Java applet, no software installation is required or no plugin is installed. The applet is just loaded temporarily onto the client`s browser. It can be used wherever is an access to the Internet. A web browser, a microphone, and a headset; sufficient to make calls.

It is completely an independent application. Telesis WebPhone works with all major browsers such as Google Chrome, Internet Explorer, Mozilla Firefox, Opera, or Safari. The WebPhone uses the Java Runtime Environment (JRE) 6.0 or higher that has been preinstalled on almost every computer for several years. Older computers can be updated free from Java site.

Performance and Optimization

The implemented protocol is the industry standard Session Initiation Protocol. However, Telesis IP Telephony systems, which host the applet, have been optimized in many respects like optimum frame lenghts, advanced echo cancellation, adaptive jitter buffering. Especially, the jitter buffer in Telesis systems is made intelligent and adaptive for the received RTP packets from the client side. Thanks to this, the best audio performance is ensured wherever the call comes from.

Easy Web Site Deployment

Telesis WebPhone feature empowers web sites of Telesis system owners. No need to have a special experience. It is so easy. Telesis system owners may just place a single and simple html link from their personal or commercial web sites to their Telesis systems. Hence, deploying webphone service on a web site takes a few minutes only. But, the result is an attractive web site allowing clients click to make calls. You may click on this link to access Telesis Webphone, which is deployed in this web site and hosted in Telesis IP PBX system in our premises.

Advanced Web Site Deployment

Web site designers may prefer to prepare their own personal or commercial web pages with names and corresponding click to call buttons. Then, click to call buttons may be individually linked to the hosting Telesis system. It is also very easy. A little experience in web design may result in fancy web pages.

Direct Web Dialing In: Calling Analog, Digital or IP Users of a Telesis System

Telesis system may also be enabled for DWDI (Direct Web Dialing In) service.  Thus, any client, who are aware of the number to be accessed, may just edit this number in the address bar of the web browser. This number, which may be any type of user (analog, digital or IP), is called in the Telesis system.

Email Signatures with Web Call Invitation

Telesis system users may add their web call links in their email signatures. Consequently, email receipents may just click on this link to call the sender.

WebPhone Update

Since the Telesis WebPhone is a temporary applet that is loaded from the Telesis system with each session, clients do not need to check any updates. All SIP and other parameters are inserted into the java applet automatically by the hosting Telesis system. Consequently, clients do not have to dial anything or to make any settings. No registration is required. Application is just a click to call one. These are good features, especially for non-technical people.  If there is an updated version available in Xymphony operating firmware of the hosting Telesis system, clients will be automatically using new features (if any) with their next calls. 

 

 

Miscellaneous Services and Features

 

 

---

 

Dynamic DNS Update

Telesis Systems Offering Dynamic DNS Update Client Service

• Telesis Business Phone Systems:
  • PX24N (Telesis Nano) IP PBX SoHo System
  • PX24U Hybrid IP PBX Business Phone System
  • PX24M Hybrid IP PBX Business Phone System
  • PX24X Hybrid IP PBX Business Phone System
• Telesis Switching Systems:
  • X1 Large Capacity TDM - IP Telephony Switch
• Combined VoIP Gateway and Signaling Converters:
  • Stillink 200
  • Stillink 800
  • Stillink 3200

Description

Dynamic DNS update client service is especially useful for a Telesis system with the dynamic public IP address, which changes periodically (monthly, weekly, or daily). It allows the system administrator to make sure he can always access a Telesis system from anywhere in the world with using its registered hostname (like mytelesis.dyndns.org) without worrying about its dynamic public IP address. Furthermore, VoIP networking Telesis systems with dynamic IP addresses becomes easy with the availability of this service.

 

---

 

XDP - Xymphony Discovery Protocol

Telesis Systems Offering XDP

• Telesis Business Phone Systems:
  • PX24N (Telesis Nano) IP PBX SoHo System
  • PX24U Hybrid IP PBX Business Phone System
  • PX24M Hybrid IP PBX Business Phone System
  • PX24X Hybrid IP PBX Business Phone System
• Telesis Switching Systems:
  • X1 Large Capacity TDM - IP Telephony Switch
• Combined VoIP Gateway and Signaling Converters:
  • Stillink 200
  • Stillink 800
  • Stillink 3200
• IP Phones
  • Telesis ITS821 Executive IP Telephone Set

Description

Xymphony Discovery Protocol (XDP) is a discovery protocol that has been integrated into Telesis PX24, X1, Stillink IP systems and Telesis ITS821 IP telephones.

XDP allows system servicemen and administrators to quickly and easily locate Telesis IP systems and IP phones in a network. XDP is especially useful when maintaining Telesis systems in a multi-vendor network environment, which has a limited information only.

Freeware Telesis XTools Utility may discover Telesis IP systems and IP phones in a network easily and quickly

With multicasting, the freeware Telesis XTools Utility delivers discovery information to all Telesis IP systems and IP phones simultaneously. Accordingly, these may return with their:

• Mac addresses, 
• IP addresses, 
• Web server port numbers, 
• Paltform types (PX24U, PX24M, PX24X, Stillink, ...),
• Subnet masks,
• Gateway addresses,
• Easy setup status (enabled or disabled),
• Xymphony or firmware versions,
• Given system names,

etc.

Furthermore, XDP brings some other functionalities to Telesis IP systems like easy programming of the system settings and resolving IP address conflicts.