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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 platform OpenScape Business.

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

Overview

OpenScape Business has 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 SIP Providers.

SIP trunking features

• Up to 8 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

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

• 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 all providers. These connections must be implemented via TDM trunks.
• For DTMF transmission outband transport via RFC 2833/RFC4733 is highly recommended. The ITSP MUST support outband DTMF (RFC2833) if UC Suite functions via SIP trunk (e.g. auto attendant, voicemail control) shall be used.
• For DTMF transmission inband transport may be used, but the functionality cannot be guaranteed as endpoints (e.g. UC-Suite) may not support inband DTMF.
• For Fax transmission the T.38 protocol is highly recommended. From OpenScape Business V2R0.3 UC Suite is able to proceed T.38 and G.711 faxes as well.
• If your provider does not support Fax T.38, Fax pass-through with G.711 is possible but transmission quality depends on network infrastructure in this case.
• CLIP no screening (set a different number to be displayed at called party) is available in OpenScape Business from V1R3 with ITSPs in DID mode. More details in "OSBiz V1/V2 Configuration for ITSP".
• In a network, a configured ITSP can be used at each respective node. From OpenScape Business V1R3 only one centralized ITSP in a network can be used.(with ITSP in DID mode)
• An incoming ITSP call can be transferred or forwarded to another public subscriber via ITSP. Concatenation of such 'trombone scenarios' is not supported. So 2 lines (one incoming and one outgoing) are used while the call is active.

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.

1. 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.
2. Get the IP and SIP settings for the specific provider. Follow the instructions listed in the "VoIP Provider Data Collection" document and send the profile to the VoIP Provider certification team. This is essential for correct PBX setup.
     VOIP provider data collection and test list - last update: 2017-02-08

--> Hint: Simplified and optimized test procedure from OSBiz V2.

1. 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.
2. Deliver the test results, including all Wireshark traces and final provider data to the VoIP Provider certification team.
3. Prepare a Provider specific configuration guide, which will be published on this wiki page later on. An example-doc is included in the ZIP-file. Use English screenshots from OpenScape Business please. Explanations can be added in your language. e.g. German Provider in German language, French Provider in French language, International provider in English language... Deliver the document 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 OpenScape Business administration for easy setup.

General Information by platform

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

OpenScape Business

• For common features and restrictions see overview
• Concurrent calls via ITSP (depending on the available bandwidth and used Codec).
  • 30 for OpenScape Business X1
  • 60 for OpenScape Business X3 / X5 / X8
  • 180 for OpenScape Business S
• An external router is always necessary for OpenScape Business S, OpenScape Business X1 / X3 / X5 / X8 can optionally be operated with the bulit-in router / firewall.
• Mobility Entry with DTMF call control is supported for OpenScape Business X1 / X3 / X5 / X8, S.
• FAX over IP with T.38 is supported for analog Fax devices, terminal adapters and the OpenScape Business UC Suite.
• *If your provider does not support Fax T.38, Fax pass-through with G.711 is possible but transmission quality depends on network infrastructure in this case.
• From OpenScape Business V2R0.3 UC Suite is able to proceed T.38 and G.711 faxes as well.
• 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.

General Configuration guides

  OSBiz V1 Configuration for ITSP- latest update: 2014-12-17

  OSBiz V2 Configuration for ITSP- latest update: 2016-01-15

  HowTo-CLIPnoScreeningEng- latest update: 2015-10-26

  HowTo-CLIPnoScreeningDeu- latest update: 2015-11-04

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 Overview about Released SIP Providers below.

International

VoIP Providers	Configuration Guide
COLT	-
Verizon	AppNote OpenScape_Business_V1 with_Verizon_IP_Trunks ML 10052013 V05
Skype Connect	-

• 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

Austria

Belgium

Brazil

Canada

Czech Republic

Chile

Denmark

Finland

France

Germany

VoIP Providers	Configuration Guide
1&1	-
Vodafone (formerly Arcor)	-
Anlagenanschluss R3 (DID)	Vodafone IP-ALA-R3 Germany latest update: 2016-11-04
ennit AG(formerly tng)	-
GMX (now 1&1)	-
Purtel	-
QSC IPfonie extended (DID)	-
QSC IPfonie extended Connect (DID)	-
sipgate basic (MSN)	-
sipgate trunking (DID)	-
toplink	-
Deutsche Telekom IP-As mit MSN	Deutsche Telekom DIPVD latest update: 2016-06-03
Deutsche Telekom DeutschlandLAN SIP-Trunk	Deutsche Telekom ALL-IP SIP-Trunk latest update: 2016-09-07
ecotel sipTrunk DDI	-
ecotel sipTrunk Connect 1.0	-
ecotel sipTrunk 2.0	Ecotel_SipTrunk2.0_Germany latest update: 2017-01-11
M-Net	-
MK Netzdienste	-
Peoplefone AG	-
Equada	-
HFO	-

Greece

VoIP Providers	Configuration Guide
CosMote	COSMOTE Greece latest update: 2016-10-02

Italy

Liechtenstein

VoIP Providers	Configuration Guide
/ FL1 Telekom)	FL1 Liechtenstein latest update: 2016-04-14

Mexico

The Netherlands

VoIP Providers	Configuration Guide
Infopact	-
OneCentral	-
OneXS	-
Priority Telecom	-
RoutIT	-
SIGNET B.V.	SIGNET telecom Netherlands latest update: 2016-09-30
Speakup	-
Tele2	-
Vodafone NL	-
Voiceworks	-
Ziggo	-
T-Mobile NL	-
Motto Communications	-
BCOM	BCOM Netherlands latest update: 2016-03-21
X2COM	X2COM Netherlands latest update: 2016-06-03
BLU	OpenScape Business V2 Config ITSP BLU telecom latest update: 2016-05-02
KPN	-

New Zealand

Portugal

Serbia

Slovakia

Slovenia

South Africa

Spain

VoIP Providers	Configuration Guide
LCR (Least Cost Routing)	-
vozelia	-
Xtra Telecom	-
Telecable	-
VozTelecom	VozTelecom_Spain latest update: 2016-08-07

Sweden

Switzerland

VoIP Providers	Configuration Guide
Cablecom	-
efon	-
Sunrise	Sunrise_SIP-Trunk_Swiss latest update: 2017-01-11
Swisscom BCON	-
vtx	-
Telco Pack SA	-
Peoplefone AG	-
Swisscom Smart Business Communication	-
Swisscom VoIP Gate	-
Swisscom Enterprise SIP	-

United Kingdom

VoIP Providers	Configuration Guide
BT IPVS	-
Gamma	-
HIPCOM	-
Opal	-
Voiceflex	-
VoIP Ltd i-Line	-
tIPicall	-
TalkTalk	-
KCOM	KCOM UK ITSP SIP Trunk Configuration latest update: 2016-06-03

United States of America

VoIP Providers	Configuration Guide
AT&T	OSO_ATT SipTrk Configuration Note 07092013 V1
Cbeyond	-
CenturyLink	-
Sotel	-
Windstream   (Broadsoft platform only)	-
XO	-
Intermedia	-

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 about Released SIP Providers Latest update: 2017-01-11

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

• Partner Portal (login required, Internet Explorer only)
•   VoIP Anbieter (Übersicht, Feature, Vergleiche)