Difference between revisions of "Collaboration with VoIP Providers"

Revision as of 14:11, 24 February 2010

Collaboration with VoIP Providers (also known as ITSP) via SIP Trunking is an important topic in the todays world of Voice/Data Communication over the Internet. The following information will give an overview of the supported VoIP providers in the different countries as well as additional considerations and hints to the connected platforms.

You did not find your provider? Find out how to get it released!
.

Overview

The SME platforms have been tested with a large number of Internet telephony providers that support SIP. Continuously new providers are tested, released and included within administration. As the SIP recommendations leave some room for interpretation, there are differences in the range of features supported with a specific provider. In this chapter general information is given which is valid for all platforms and all SIP Providers.

SIP trunking features for all SME platforms

• up to 4 active SIP providers and one Internet connection
• direct connection to DSL modem or operation behind another router using STUN
• SIP trunking with single numbers (MSN) and up to 30 SIP user accounts
• Direct Inward Dialing (DID)
• most important features available
  • CLIP
  • Hold, Consultation, Toggle
  • Transfer (attended, semi-attended)
  • Ringing group, Call pickup
  • Conference
  • Call Forwarding
  • DTMF transmission e.g. for voicemail access
• number of simultaneous VoIP connections is only limited by DSL bandwidth
• automatic fallback to ISDN at SIP trunk failure
• Voice Codecs G.711, G.729 and G.723
• Fax over IP via T.38 (or G.711)

Interoperability with Carrier VoIP Platforms

The SME platforms with SIP Stack 4.0.26 have been validated with the BroadWorks SIP Interface. BroadWorks is the Carrier VoIP solution from Broadsoft, the global market leader.

•   Broadsoft Validation
•   Broadsoft Partner Configuration Guide

General Requirements for VoIP

• LAN with 10/100/1000 MBit/s and no more than 40% network load
• Separate port on the switch or router for every component in the IP network (no hubs as concentrators)
• Not more than 50 ms delay in one direction (One Way Delay); not more than 150 ms total delay
• Not more than 3% packet loss and 20 ms jitter
• Quality of Service (QoS) support - IEEE 802.p, DiffServ (RFC 2474) or ToS (RFC 791)
• Sufficient WAN bandwidth (uplink and downlink) for intended simultaneous calls and CODEC
• External modem, such as a DSL modem
• Intermediate routers shall support quality of service and bandwidth control mechanisms if the WAN link is used for voice and data simultaneously. SIP messages have to be routed transparently.

Restrictions for all Providers and SME Platforms

• Special call numbers and emergency calls
  • Special call numbers (e.g. 0800, 0900, 0137) or analog fax/modem connections are not supported by all SIP providers.
  • Emergency call numbers (e.g. 110, 112) are not supported by most providers. These connections must be implemented via S0 access (administration default)
• Only one external destination is allowed for ringing groups which are to be called via SIP trunk from VoIP provider.
• SIP trunks can not be used for dialing through or direct inward system access with DTMF tones e.g for the feature Mobility Entry. Please use a TDM line for those features.
• DTMF is supported via RFC 2833 or inband. SIP INFO method is not supported.
• For Fax transmission the T.38 protocol is highly recommended. Fax pass through using codec G.711 has been found much less reliable in most SIP provider environments. If your provider does not yet support Fax T.38, Fax pass through with G.711 is possible but there is no warranty or further support.

How to get a new VoIP provider released

Tests are necessary with every new provider before making installations in the field due to different implementations of the SIP protocol.

• First the VOIP / SIP trunking providers which are most important for your business are selected.
• After that the VoIP provider test list is worked through in the country according to the instructions included. This covers the most important features at the SIP trunking interface. Leading switch should be the OpenScape Office MX, after that a reduced test list can be used for HiPath 3000.
• Remote support is available by the VoIP Provider cerification team. The final test results including traces and provider data shall be sent to the VoIP Provider cerification team (karl-werner.weigt@unify.com).

  VOIP provider testlist

• The appropriate VoIP provider template and profile will be included in the HiPath / OpenScape Office administration for easy setup.
• After successful tests the VoIP provider is released, and published on this page. Back level support is available for the new provider after that.

General Information by platform

For discontinued platforms see Collaboration with VoIP Providers (discontinued platforms).

HiPath 3000

HiPath 3000 ≥V7 with HG 1500:

• For common features see overview
• An external router is always necessary
• Maximum 30 concurrent calls between IP phones and ITSP (depending on the available bandwidth and used CODEC). For TDM phones 1 B-channel at HG 1500 has to be available for each concurrent call.
• No FAX via ITSP from/to OpenScape Office client applications. A TDM trunk has to be used.
• SIP trunking is not released for use with the OpenScape Office Contact center application.

HiPath OpenOffice ME

SIP Trunking features

• See overview.
• G.711 and G.729 codecs are supported by the HiPath OpenOffice appliance.
• For Call Forwarding to SIP provider announcements before connect are not audible (solved with V2).
• FoIP (FAX over IP) with T.38 is available since the release of HiPath OpenOffice ME V1 R4 for FAX devices connected to Analog ports or Terminal Adapters. For FAX communication with OpenScape Office (OSO) a TDM Trunk has to be used.
  • The following T.38 FAX scenarios via ITSP are supported by HiPath OpenOffice ME:
    • 1. FAX devices (G.3) connected to an a/b (Analog) subscriber port of an HiPath GMSA, GMAA or GMAL Module or FAX devices (G.4) connected to a BRI/S0 (Digital) subscriber port of an HiPath GMSA or GMS Module.
    • 2. FAX devices (G.3) connected to an Analog port of an HiPath AP1120 Terminal Adapter (with SIP Loadware).
  • Currently not supported:
    • FAX services via ITSP from/to OpenScape Office client applications.

Configuration

HiPath OpenOffice ME V1 Configuration Guide for Internet Telephony:

•   HiPath OpenOffice ME Config Guide ITSP

OpenScape Office MX

SIP Trunking features

• All features, requirements and restrictions listed in overview are valid.

further details will be provided soon

Tested VoIP Providers by Countries

The main SIP trunking functions of the SME platforms have been tested with the following Internet telephony providers. For detailed information see the document Released SIP Providers below.

= has been successfully tested by development or customer
DID = accounts for PBX-trunking / Direct Inward Dialling
MSN = accounts for single number(s)

International

VoIP Providers	Type	HiPath 3000	HiPath OpenOffice ME	OpenScape Office MX
COLT	DID

Please check provider home page for supported countries.

Austria

VoIP Providers	Type	HiPath 3000	HiPath OpenOffice ME	OpenScape Office MX
UPC / inode	MSN
WNT	DID

Belgium

VoIP Providers	Type	HiPath 3000	HiPath OpenOffice ME	OpenScape Office MX
Belgacom
Hexacom

Brazil

VoIP Providers	Type	HiPath 3000	HiPath OpenOffice ME	OpenScape Office MX
CTBC
GVT

Czech Republic

VoIP Providers	Type	HiPath 3000	HiPath OpenOffice ME	OpenScape Office MX
Telefonica O2

Finland

VoIP Providers	Type	HiPath 3000	HiPath OpenOffice ME	OpenScape Office MX
Elisa

France

VoIP Providers	Type	HiPath 3000	HiPath OpenOffice ME	OpenScape Office MX
OpenIP
Optimitel

Germany

VoIP Providers	Type	HiPath 3000	HiPath OpenOffice ME	OpenScape Office MX
1&1	MSN
Arcor	DID
freenet	MSN
freenetBusiness	DID
GMX	MSN
Purtel	MSN
QSC	DID
sipgate	MSN DID
T-Online (discontinued)	MSN
toplink	MSN DID
XSip	DID

Italy

VoIP Providers	Type	HiPath 3000	HiPath OpenOffice ME	OpenScape Office MX
Infracom (ex Multilink)

The Netherlands

VoIP Providers	Type	HiPath 3000	HiPath OpenOffice ME	OpenScape Office MX
I2Connect	MSN DID
Infopact	MSN DID
OneCentral	MSN DID
Priority Telecom	MSN DID

Portugal

VoIP Providers	Type	HiPath 3000	HiPath OpenOffice ME	OpenScape Office MX
Portugal Telecom

Slovenia

VoIP Providers	Type	HiPath 3000	HiPath OpenOffice ME	OpenScape Office MX
Amis	MSN DID
Detel	MSN DID
MegaM	MSN DID
tustelekom	DID
T-2	DID
UPC Telemach	MSN DID

South Africa

VoIP Providers	Type	HiPath 3000	HiPath OpenOffice ME	OpenScape Office MX
Telfree	MSN DID

Spain

VoIP Providers	Type	HiPath 3000	HiPath OpenOffice ME	OpenScape Office MX
vozelia	MSN DID
Xtra Telecom	MSN DID

Sweden

VoIP Providers	Type	HiPath 3000	HiPath OpenOffice ME	OpenScape Office MX
Telia	MSN

Switzerland

VoIP Providers	Type	HiPath 3000	HiPath OpenOffice ME	OpenScape Office MX
Cablecom
sipcall	MSN
SofiCall	MSN
vtx	DID

United Kingdom

VoIP Providers	Type	HiPath 3000	HiPath OpenOffice ME	OpenScape Office MX
Gamma	DID
Opal	DID
Voiceflex	DID

United States of America

VoIP Providers	Type	HiPath 3000	HiPath OpenOffice ME	OpenScape Office MX
Cbeyond
Sotel	MSN DID

Released SIP Providers in Detail

The table provided here shows the ITSP providers that were successfully checked in a connectivity test. More details about test results are given in the pdf-file Overview of released SIP providers below. Informations concerning CLIP, CLIR, COLP etc. are listed, restrictions may be reported as well.

•   Overview of released SIP providers Latest update: 2010-02-24

Connectivity Test Information

The connectivity test is carried out with

• HiPath 3000
• OpenScape Office MX

Tests are done for MSN- and DID-account (as long as the provider´s realm is different), also the platforms are checked as NAT router and / or behind a router.

Several test scenarios are clustered in the test list, main issues are:

• Registration
• Basic Calls from / to ITSP
• Feature Tests
  • Consultation, transfer, toggle, conference
  • Call Pickup / ringing group scenarios
  • call forwarding scenarios
  • DTMF
  • CLIP / CLIR
  • COLP / COLR
  • FAX
• Special Tests
  • emergency calls
  • service calls
  • long duration calls

The connectivity tests are performed using the following endpoints:

• Analogue
• ISDN
• HFA-phones
• UP0 phones
• SIP-phones
• AP 1120 HFA / SIP
• optiClient 130 HFA
• WLAN2-phones HFA / SIP

At least 3 different types of phones are used for each test case.

Frequently asked Questions

• VoIP Provider FAQ - answers to frequently asked questions

External Links

• Siemens Enterprise Business Area (SEBA) (login required, Internet Explorer only)
•   VoIP Anbieter (Übersicht, Feature, Vergleiche)