Collaboration with VoIP Providers

VoIP Providers released for Unify SME Platforms > table

The connection of communication systems to the public network via SIP Trunking can be used instead of or in addition to traditional PSTN trunks. Many VoIP Providers, also known as ITSPs, are offering corresponding services. The following article gives an overview of the SIP trunking features and limitations and lists the Providers, which have been tested and released with 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 (can be configured per provider with current software) - integrated SBC functionality
SIP trunking with single numbers (MSN) and up to 30 SIP user accounts (called Internet telephony stations in WBM)
Direct Inward Dialing (DID)
Comprehensive feature set available e.g.
- CLIP
- Hold, Consultation, Toggle
- Transfer (attended, semi-attended)
- Ringing group, Call pickup
- Conference
- Call Forwarding
- DTMF transmission e.g. for voicemail access
Simultaneous VoIP connections depending on system and 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 SIP Endpoint Configuration

The following changes have been made starting from HiPath 3000 V9 and OpenScape Office V3R2:

Enhanced SIP attack protection has been implemented. For diagnostics and SIP provider configuration hints see How to enable blocked SIP Providers.
STUN handling has been improved. STUN can now be configured per ITSP profile. If you were not using STUN for a provider (Global STUN mode is off), please verify that the profile flag "Use STUN" is disabled. If you were using STUN (Global STUN mode is other than off), then please verify that the flag "Use STUN" is enabled in the provider profile. For further information see the ITSP configuration guides at HiPath 3000 or OpenScape Office.

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 tested SIP Stack is used in HiPath 3000 V9 and OpenScape Office V3 as well. Broadsoft Validation

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.

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 many providers. These connections must be implemented via S0 access (administration default).
DTMF is supported outband via RFC 2833 or inband. The ITSP has to support outband DTMF (RFC2833) for using the Autoattendant via SIP trunk.
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.
CLIP no screening (set a different number to be displayed at called party) is not available with our platforms for SIP trunks. Please check with your ITSP, if this can be accomplished at the provider side.
In a network, a configured ITSP can only be used at each respective node not network-wide.
Having two calls from one subscriber via ITSP trunk is possible e.g. for call forwarding or for call transfer. Concatenation of such 'trombone scenarios' is not supported.

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. Otherwise no support is available for an ITSP. Test resources have to be provided by the region or partner.

Select the VOIP / SIP trunking providers, which are most important for your business, estimate business relevance and get agreement for start of certification from product management.
Get the IP and SIP settings for the specific provider. Send the completed document "VoIP Provider Data Collection" to the VoIP Provider certification team. This is essential for correct PBX setup. The tables in chapter 2 have to be filled in.
VOIP provider data collection and test list - last update: 2013-06-19
Perform all tests in the VoIP provider test list from above, according to the instructions included. System configuration and tests require trained local personal. Remote support is available by the VoIP Provider certification team.
Deliver the test results, including all Wireshark traces and final provider data to the VoIP Provider certification team.

After successful tests the VoIP provider is released, and published on this page. Regular back level support is available for issues with the new provider after that. A preconfigured VoIP provider profile will be included in the HiPath / OpenScape Office administration for easy setup.

General Information by platform

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

OpenScape Business

The SIP trunking interface of OpenScape Business V1 is backward compatible with HiPath 3000 and OpenScape Office. All tested VoIP Providers are supported.

For common features and restrictions see overview
Concurrent calls via ITSP (depending on the available bandwidth and used Codec).
- 60 for OpenScape Business X3 / X5 / X8
- 128 for OpenScape Business S
An external router is always necessary for OpenScape Business S, OpenScape Business X3 / X5 / X8 can optionally be operated with the bulit-in router / firewall.
Mobility Entry with DTMF call control is supported for OpenScape Business X3 / X5 / X8, but not for OpenScape Business S.
FAX over IP with T.38 is supported for analog Fax devices, terminal adapters and the OpenScape Business UC Suite. Fax G.711 is not supported via the UC Suite.
A trunk prefix to select a SIP trunk as an alternate trunk can be used from the OpenScape Business client's dial box but not from client's directories and settings (myPortal, myAttendant, myAgent).
SIP trunks cannot be used for presence based call forwarding, if a TDM trunk is also present because of the limitation above.

Configuration

OpenScape Business Configuration Guide for Internet Telephony:

OSBiz V1 Configuration for ITSP

HiPath 3000

Valid for HiPath 3000 V9 with HG 1500.

SIP Trunking features

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.
For OpenScape Office HX related features Fax via application, trunk prefix use and presence based forwarding see below.

Configuration

HiPath 3000 Configuration Guide for Internet Telephony:

HiPath 3000 Config Guide ITSP

OpenScape Office

Valid for OpenScape Office V3 LX / MX / HX.

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 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. 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.
A trunk prefix to select a SIP trunk as an alternate trunk can be used from the OpenScape Office client's dial box but not from client's directories and settings (myPortal, myAttendant, myAgent).
SIP trunks cannot be used for presence based call forwarding, if a TDM trunk is also present because of the limitation above.

Configuration

OpenScape Office Configuration Guide for Internet Telephony:

OpenScape Office Config Guide ITSP

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	OpenScape Office	OpenScape Business
COLT	DID
Skype Connect 3)	MSN

Please check provider home page for supported countries.
Skype Connect 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.

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

Australia

VoIP Providers	Type	HiPath 3000	OpenScape Office	OpenScape Business
Engin	MSN
Internode	MSN

Austria

VoIP Providers	Type	HiPath 3000	OpenScape Office	OpenScape Business
UPC / inode 3)	MSN		-
IP Austria / my Tweak 3)	DID

Belgium

VoIP Providers	Type	OpenScape Office
Belgacom 3)	DID
Belgacom IMS 3)	MSN / DID	*)
Destiny	MSN / DID
Hexacom 3)	MSN / DID	-
Telenet	MSN / DID
WIN 3)	MSN / DID	*)

 *) Platform released, OpenScape Office application test pending

Brazil

VoIP Providers	Type	HiPath 3000	OpenScape Office
CTBC 1)	DID		-
GVT 1)	DID
Voitel	MSN	-

Czech Republic

VoIP Providers	Type	HiPath 3000	OpenScape Office	OpenScape Business
GTS	DID
T-mobile	MSN / DID
Telefonica O2	MSN / DID

Chile

VoIP Providers	Type	HiPath 3000	OpenScape Office	OpenScape Business
Entel NGN	MSN		-

Denmark

VoIP Providers	Type	HiPath 3000	OpenScape Office	OpenScape Business
Uni-tel A/S	MSN / DID

Finland

VoIP Providers	Type	HiPath 3000
Elisa	DID
Länsilinkki	MSN	-
Telia-Sonera	DID
TDC	MSN / DID

France

VoIP Providers	Type	HiPath 3000
Acropolis Telecom 3)	DID	-
OpenIP 3)	DID	-
Paritel Operateur	MSN / DID
Completel	DID	-

Germany

VoIP Providers	Type	OpenScape Office
1&1 3)	MSN	-
Vodafone (formerly Arcor) 3)	VoIP Anlagenanschluss (DID)
ennit AG(formerly tng)	DID / MSN
GMX (now 1&1) 3)	MSN	-
Purtel 3)	MSN	-
QSC	IPfonie extended (DID)
sipgate 3)	sipgate basic (MSN) sipgate trunking (DID)
toplink 3)	MSN / DID

Italy

VoIP Providers	Type	HiPath 3000	OpenScape Office
Infracom (ex Multilink) 3)	MSN		-
TWT	MSN / DID	-
Viatek	MSN / DID

Mexico

VoIP Providers	Type	HiPath 3000	OpenScape Office	OpenScape Business
Telefonica Mexico 1)	MSN	-

The Netherlands

VoIP Providers	Type	HiPath 3000	OpenScape Office	OpenScape Business
Infopact 3)	MSN / DID
OneCentral 3)	MSN / DID
OneXS	MSN / DID
Priority Telecom 3)	MSN
RoutIT	DID
Speakup 3)	MSN / DID
Tele2	MSN / DID
Vodafone NL	MSN / DID
Voiceworks 3)	DID

New Zealand

VoIP Providers	Type	HiPath 3000	OpenScape Office	OpenScape Business
TelstraClear 3)	MSN / DID

Portugal

VoIP Providers	Type	HiPath 3000	OpenScape Office	OpenScape Business
Portugal Telecom	DID

Slovakia

VoIP Providers	Type	HiPath 3000	OpenScape Office	OpenScape Business
Slovak Telekom 3)	DID

Slovenia

VoIP Providers	Type	HiPath 3000
Amis 3)	MSN
Detel 3)	MSN
MegaM 3)	MSN
tustelekom 3)	DID
T-2 3)	DID	-
Telemach 3)	MSN

South Africa

VoIP Providers	Type	HiPath 3000
Nashua ECN	MSN / DID
Telfree 3)	MSN / DID	-
Vodacom	MSN / DID

Spain

VoIP Providers	Type	OpenScape Office
LCR (Least Cost Routing)	MSN / DID
vozelia 3)	MSN / DID	-
Xtra Telecom 3)	MSN

Sweden

VoIP Providers	Type	HiPath 3000	OpenScape Office	OpenScape Business
Tele2 3)	DID
Telia 3)	MSN

Switzerland

VoIP Providers	Type	HiPath 3000
Cablecom	DID
sipcall	MSN	-
SofiCall 3)	MSN	-
vtx 3)	DID

United Kingdom

VoIP Providers	Type	HiPath 3000	OpenScape Office	OpenScape Business
BT IPVS	MSN / DID
Gamma 3)	DID
HIPCOM	MSN / DID
Opal 3)	DID
Voiceflex 3)	DID

United States of America

VoIP Providers	Type	HiPath 3000	OpenScape Office
AT&T	DID	-	OSO_ATT SipTrk Configuration Note 07092013 V1
Cbeyond 3)	MSN		-
CenturyLink 2)	DID		-
Sotel 3)	MSN / DID
Windstream 3) (Broadsoft platform only)	MSN / DID
XO 1)	MSN / DID	-

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.
2) No fixed servers (manual config needed), two profiles with different server addresses and identical accounts and MSN shall be used.
3) ITSP does not support Fax T.38. Fax communication with OpenScape Office application / OpenScape Business UC Suite is not possible. Other Fax over IP communication may be unreliable.

Released SIP Providers in Detail

The table above shows the ITSPs that passed successfully a connectivity test. More details about test results, supported features and restrictions for a specific ITSP are listed in following PDF document.

Overview of released SIP providers 20130702 Latest update: 2013-07-02

The following scenarios are covered within the connectivity test:

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
Calls handled via UC application

The connectivity tests are performed using various endpoints (for details see test list).

Frequently asked Questions

VoIP Provider FAQ - answers to frequently asked questions

External Links

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

Views

Collaboration with VoIP Providers

The Wiki of Unify contains information on clients and devices, communications systems and unified communications. - Unify GmbH & Co. KG is a Trademark Licensee of Siemens AG.

Contents

Overview

SIP trunking features for all SME platforms

General Requirements for VoIP

Restrictions for all Providers and SME Platforms

How to get a new VoIP provider released

General Information by platform

OpenScape Business

HiPath 3000

OpenScape Office

Tested VoIP Providers by Countries

International

Australia

Austria

Belgium

Brazil

Czech Republic

Chile

Denmark

Finland

France

Germany

Italy

Mexico

The Netherlands

New Zealand

Portugal

Slovakia

Slovenia

South Africa

Spain

Sweden

Switzerland

United Kingdom

United States of America

Released SIP Providers in Detail

Frequently asked Questions

External Links

Personal tools

Information

Helpdesk

Wiki Internal

Tools