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Difference between revisions of "Collaboration with VoIP Providers"

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[[Image:VoIP Provider SME.jpg|thumb|400px|VoIP Provider SME.jpg]] '''Collaboration with VoIP Providers''' (also known as [[ITSP]]) via [[SIP Trunking]] is an important topic in the todays world of IP Voice/Data Communication. The following information will give an overview of the supported [[VoIP]] Providers in the different countries as well as additional considerations and hints regarding the SME platforms.  
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[[Image:VoIP Provider SME.jpg|thumb|450px|VoIP Provider SME.jpg]] '''Collaboration with VoIP Providers''' (also known as [[ITSP]]) via [[SIP Trunking]] is an important topic in the todays world of IP Voice/Data Communication. The following information will give an overview of the supported [[VoIP]] Providers in the different countries as well as additional considerations and hints regarding the SME platforms.  
 
  
 
'''You did not find your provider? [[#How_to_get_a_new_VoIP_provider_released|Find out how to get it released!]]''' <br>.  
 
'''You did not find your provider? [[#How_to_get_a_new_VoIP_provider_released|Find out how to get it released!]]''' <br>.  

Revision as of 19:20, 21 April 2011

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Icon linkarrow.gif Home  Icon linkarrow.gif Communications Systems  Icon linkarrow.gif Collaboration with VoIP Providers
File:VoIP Provider SME.jpg
VoIP Provider SME.jpg
Collaboration with VoIP Providers (also known as ITSP) via SIP Trunking is an important topic in the todays world of IP Voice/Data Communication. The following information will give an overview of the supported VoIP Providers in the different countries as well as additional considerations and hints regarding the SME 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
  • operation behind another router using STUN
  • SIP trunking with single numbers (MSN) and up to 30 SIP user accounts (called Internet telephony stations in WBM)
  • 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
  • large number of simultaneous VoIP connections depending on available DSL bandwidth, used Codec and system. G.711 needs up to 92 kbit/s, G.729A needs up to 36 kbit/s in both directions.
  • automatic fallback to ISDN at SIP trunk failure
  • Voice Codecs G.711 and G.729
  • Fax over IP via T.38 (or G.711)


Security Warning - Malicious attacks from the Internet may lead to reduced service quality or toll fraud!
     Please strictly follow these basic rules:
  • Never open up the firewall in an internal or external router e.g. by forwarding port 5060.
    The SME platforms take care of opening the firewall for traffic with the selected VoIP providers. Attacks from other internet addresses are therefore blocked. Opening up the router's firewall would invalidate this security measure.
  • Never configure SIP subscribers at a HiPath / OpenScape system without a strong password.
    Authentication blocks registration of unauthorised SIP clients. See pdf.png  SIP Endpoint Configuration

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. The tested SIP Stack is used in the current system versions Hipath 3000 V8 and OpenScape Office V3 as well.

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.

For interworking with different network configurations and edge devices and how to use STUN please see Network Configuration for VoIP Providers. One common STUN setting must be used for all providers in a system.

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.
  • 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.
  • Other media types besides audio (voice and voice band data) and image (fax) are not supported.
  • The ITSP has to support outband DTMF (RFC2833) for using the OpenScape Office Autoattendant via SIP trunk.


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 and agreed with product management.
  • Next step is to get the IP and SIP settings necessary for interworking with the specific provider. The document "VoIP Provider Configuration" has to be completed and sent to the VoIP Provider cerification team to allow correct initial PBX setup.
  • After that, the VoIP provider test list has to be worked through in the country, according to the instructions included. Remote support is available by the VoIP Provider cerification team. The tests cover the most important features at the SIP trunking interface. Leading switch should be the OpenScape Office MX, after that a reduced delta test list can be used for HiPath 3000.

zip.png  VOIP provider test list and configuration last update: 2011-03-23

  • The final test results including traces and provider data shall be delivered to the VoIP Provider cerification team.
  • The appropriate VoIP provider configuration data 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

SIP Trunking features

valid for HiPath 3000 ≥ V7 with HG 1500:

  • For common features and restrictions see overview
  • An external router is always necessary
  • Maximum 32 concurrent calls between IP phones and ITSP (depending on the available bandwidth and used Codec). For TDM devices (analog, ISDN) 1 B-channel at HG 1500 has to be available for each concurrent call (gateway call).
  • 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.
  • No FAX via ITSP from/to OpenScape Office HX client applications. A TDM trunk has to be used.
  • SIP trunking is not released for use with the OpenScape Office HX Contact center application.

Configuration

HiPath 3000 Configuration Guide for Internet Telephony:

See also: How to enter SIP Provider Account Data

OpenScape Office MX/LX

valid for OpenScape Office V2 (MX) and V3 (MX/LX):

SIP Trunking features

  • All features, requirements and restrictions listed in overview are valid.
  • Up to 32 simultaneous calls via SIP trunking for OpenScape Office MX (depending on available DSL bandwidth and used Codec)
  • Up to 128 simultaneous calls via SIP trunking for OpenScape Office LX (depending on available DSL bandwidth and used Codec)
  • An external router is always necessary for OpenScape Office LX, OpenScape Office MX can optionally be operated with the bulit-in router / firewall.
  • SIP trunks cannot 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. Starting from OpenScape Office V3, myPortal for Mobile supports comprehensive control and UC functionality.
  • FAX over IP with T.38 is supported for analog Fax devices, terminal adapters and the OpenScape Office application. Fax via G.711 is not supported with the application. No Fax over IP is available for the OpenScape Office V2 application, a TDM Trunk has to be used.
  • A trunk prefix to select a SIP trunk as an alternate trunk can not be used from OpenScape Office Clients directories (myPortal, myAttendant, myAgent).
  • SIP trunks cannot be used for presence based call forwarding, if a TDM trunk is also present.
  • Release of SIP trunks for use with the OpenScape Office Contact Center application is under preparation for OpenScape Office V3.

Configuration

OpenScape Office Configuration Guide for Internet Telephony:

See also: How to enter SIP Provider Account Data

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.

tick.gif = 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 OpenScape Office
COLT DID tick.gif tick.gif
Skype Connect MSN tick.gif tick.gif
  • Please check provider home page for supported countries.
  • Skype Connect (former name Skype for SIP) is a fee-based business service from Skype. It allows voice communication between Skype clients and office phones using the standard SIP trunking interface. Other Skype client features like chat or video are currently not supported.

Skype Connect certified logo Our SME Platforms have been officially certified by Skype. pdf.png  Skype Connect with SME platforms

Australia

VoIP Providers Type HiPath 3000 OpenScape Office
Engin MSN tick.gif tick.gif
Internode MSN tick.gif tick.gif

Austria

VoIP Providers Type HiPath 3000 OpenScape Office
UPC / inode MSN tick.gif -
IP-austria DID tick.gif tick.gif
WNT DID tick.gif -

Belgium

VoIP Providers Type HiPath 3000 OpenScape Office
Belgacom DID tick.gif tick.gif
Hexacom MSN
DID 		tick.gif
tick.gif 		-
Telenet MSN
DID 		tick.gif
tick.gif 		tick.gif
tick.gif

Brazil

VoIP Providers Type HiPath 3000 OpenScape Office
CTBC 1) DID tick.gif -
GVT 1) DID tick.gif tick.gif

Czech Republic

VoIP Providers Type HiPath 3000 OpenScape Office
T-mobile MSN
DID 		tick.gif
tick.gif 		tick.gif
tick.gif
Telefonica O2 MSN
DID 		tick.gif
tick.gif 		tick.gif
tick.gif

Chile

VoIP Providers Type HiPath 3000 OpenScape Office
Entel NGN MSN tick.gif -

Finland

VoIP Providers Type HiPath 3000 OpenScape Office
Elisa DID tick.gif -

France

VoIP Providers Type HiPath 3000 OpenScape Office
OpenIP DID - tick.gif
Optimitel DID tick.gif -

Germany

VoIP Providers Type HiPath 3000 OpenScape Office
1&1 MSN tick.gif -
Vodafone (formerly Arcor) VoIP Anlagenanschluss (DID) tick.gif tick.gif
freenet (now 1&1) MSN tick.gif -
freenetBusiness (discontinued) DID tick.gif tick.gif
GMX (now 1&1) MSN tick.gif -
Purtel MSN tick.gif -
QSC IPfonie extended (DID) tick.gif tick.gif
sipgate sipgate basic (MSN)
sipgate trunking (DID) 		tick.gif
tick.gif 		tick.gif
tick.gif
toplink MSN
DID 		tick.gif
tick.gif 		tick.gif
tick.gif
XSip (discontinued) DID tick.gif -

Italy

VoIP Providers Type HiPath 3000 OpenScape Office
Infracom (ex Multilink) MSN tick.gif -
TWT MSN
DID 		tick.gif
tick.gif

Mexico

VoIP Providers Type HiPath 3000 OpenScape Office
Telefonica Mexico 1) MSN - tick.gif

The Netherlands

VoIP Providers Type HiPath 3000 OpenScape Office
I2Connect MSN
DID 		tick.gif
tick.gif 		-
Infopact MSN
DID 		tick.gif
tick.gif 		tick.gif
-
OneCentral MSN
DID 		tick.gif
tick.gif 		tick.gif
tick.gif
Priority Telecom MSN tick.gif tick.gif
Tele2 MSN
DID 		- tick.gif1)
tick.gif1)

1) Only for OpenScape Office MX. The WAN interface with PPPoE has to be used, no operation behind an external router is supported.

Portugal

VoIP Providers Type HiPath 3000 OpenScape Office
Portugal Telecom DID tick.gif tick.gif

Slovenia

VoIP Providers Type HiPath 3000 OpenScape Office
Amis MSN tick.gif tick.gif
Detel MSN tick.gif tick.gif
MegaM MSN tick.gif tick.gif
tustelekom DID tick.gif tick.gif
T-2 DID - tick.gif
UPC Telemach MSN tick.gif tick.gif

South Africa

VoIP Providers Type HiPath 3000 OpenScape Office
Telfree MSN
DID 		- tick.gif
tick.gif
Vodacom MSN
DID 		tick.gif
tick.gif 		tick.gif
tick.gif

Spain

VoIP Providers Type HiPath 3000 OpenScape Office
vozelia MSN
DID 		tick.gif
tick.gif 		-
Xtra Telecom MSN tick.gif tick.gif

Sweden

VoIP Providers Type HiPath 3000 OpenScape Office
Tele2 DID tick.gif tick.gif
Telia MSN tick.gif tick.gif

Switzerland

VoIP Providers Type HiPath 3000 OpenScape Office
Cablecom DID tick.gif tick.gif
sipcall MSN - tick.gif
SofiCall MSN - tick.gif
vtx DID tick.gif tick.gif

United Kingdom

VoIP Providers Type HiPath 3000 OpenScape Office
Gamma DID tick.gif tick.gif
Opal DID tick.gif tick.gif
Voiceflex DID tick.gif tick.gif

United States of America

VoIP Providers Type HiPath 3000 OpenScape Office
Cbeyond MSN tick.gif -
Sotel MSN
DID 		- tick.gif
tick.gif
XO 1) MSN
DID 		- tick.gif
tick.gif


1) Access parameters depend on the region within the country. If no default provider profile is supplied within administration, please use "create profile" for configuration.

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.

Connectivity Test Information

The connectivity test is carried out with

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

External Links

Retrieved from "http://wiki.unify.com/index.php?title=Collaboration_with_VoIP_Providers&oldid=31156"