| Date
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Summary
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| 03.12.2024
|
|
herweck callamar, Virtual-Call
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|
|
Vodafone Anlagenanschluss R6
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- updated w/o profile change:
|
Swisscom Enterprise SIP
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| 05.07.2024
|
|
Net4You-Voice4Biz
|
|
|
EWE Voice+ SIP Trunk, M-net Premium SIP-Trunk, Universe SIP Connect
|
- updated w/o profile change:
|
Swisscom Smart Business Communication
|
|
|
HFO Telecom NGN, Skype for Business, Swisscom BCON, Swisscom VoipGate
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| 11.03.2024
|
|
BITel Business SIP Flex, Gamma NGN (DE), Helinet, htp Business FleX smart, wittenberg-net GmbH
|
|
|
Net4You-Voice4Biz
|
|
|
easybell, EWE Voice+ SIP Trunk
|
|
|
M-net Premium SIP-Trunk, Universe SIP Connect
|
| 11.12.2023
|
|
BITel Business SIP Flex, Gamma NGN (DE), wittenberg-net GmbH
|
|
|
EWE Voice+ SIP Trunk
|
| 06.10.2023
|
|
Drei, Dstny, Komro, Magenta, RoutIT TLS, Vodafone Anlagenanschluss R5
|
|
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Helinet, htp Business FleX smart
|
|
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BT Wholesale, Deutsche Telekom CompanyFlex
|
|
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ecotel sipTrunk Connect 1.0, ecotel sipTrunk DDI, Tele 2, UPC Austria, Vodafone Anlagenanschluss, Vodafone Anlagenanschluss R3
|
| 15.12.2022
|
|
EasyFone, envia TEL, G9 Telecom, Keyyo, Orange BE SIP, Spark Voice Connect
|
|
|
KPN TLS NL, Orange BE S&F, Sunrise, Vodafone Anlagenanschluss R4
|
|
|
Orcon Ltd, TelstraClear WSIP
|
| further details:
|
https://wiki.unify.com/wiki/Collaboration_with_VoIP_Providers#Released_SIP_Providers_in_Detail
|
More than 200 VoIP Providers released for Unify SME Platforms
> tableThe 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
- integrated SBC functionality
- SIP trunking with single numbers (MSN) and up to 30 SIP user accounts (called Internet telephony stations in WBM)
- SIP trunking with 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
- Call deflection
- DTMF transmission e.g. for voicemail access
- Location information for emergency calls
- Simultaneous VoIP connections depending on available DSL bandwidth, used Codec and system.
- Automatic fallback at SIP trunk failure
- Voice Codecs G.711 and G.729
- Fax over IP via T.38
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 an OpenScape system without a strong password.
Authentication blocks registration of unauthorised SIP clients. See http://wiki.unify.com/wiki/Features_and_Configuration_of_SIP_Devices
- For diagnostics and SIP provider configuration hints see How to enable blocked SIP Providers.
|
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 router
- 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.
Hints for all Providers
- Concurrent calls via ITSP (depending on the available bandwidth and used Codec).
- 30 for OpenScape Business X1
- up to 120 for OpenScape Business X3 / X5 / X8
- 180 for OpenScape Business S
- DTMF transmission
- outband transport via RFC 2833/RFC4733 is highly recommended. The ITSP MUST support outband DTMF (RFC2833/RFC4733) if UC Suite functions via SIP trunk (e.g. auto attendant, voicemail control) shall be used.
- inband transport may be used, but the functionality cannot be guaranteed as endpoints (e.g. UC-Suite) may not support inband DTMF.
- FAX over IP
- Please check the documentation available for your ITSP which Fax transmission is preferred.
- T.38 is activated by default in OpenScape Business
- If your provider does not support Fax T.38 or T.38 transmission causes problems, Fax pass-through with G.711 is possible but transmission quality depends on network infrastructure in this case.
- Call Forwarding
- By default an incoming ITSP call is forwarded to another public subscriber via a second ITSP trunk (one incoming and one outgoing).
- Call rerouting can be used for selected ITSPs, in this case NO line is used when call is active
- Special call numbers and emergency calls may need alternate TDM connections
- Special call numbers (e.g. 0800, 0900, 0137) are not supported by all SIP providers
- analog fax/modem connections are not supported by all SIP providers.
- Emergency call numbers (e.g. 110, 112) are not supported by all providers.
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.
|
-->> 24.07.2024 New documents and templates available check the .zip file <<--
|
- 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.
- Download
VOIP provider data collection and test list
- In the .zip-file you find the
- process description updated with new parameters V3R3
- ITSP questionnaire
- HowTo guide for system setup and hints for test execution
- the testlist updated with new test description for V3R3.1
- templates for writing a configuration guide updated for V3R3.1
After successful tests the VoIP provider will be 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.
Certified providers are entitled to label their SIP Trunking Service with the OpenScape Business certified ITSP emblem
General Configuration guides
OpenScape Business SIP Trunk Configuration
Clip no Screening (CNS) for DID SIP Trunks
Mobile Extension (MEX) Connectivity
Multisite Management for DID SIP Trunks
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
- Please check provider home page for supported countries.
Australia
Austria
Belgium
Brazil
Canada
| VoIP Providers
|
Configuration Guide
|
| BabyTEL
|
-
|
Chile
| VoIP Providers
|
Configuration Guide
|
| ENTEL NGN
|
-
|
Czech Republic
Denmark
Estonia
Finland
France
Germany
Greece
Hong Kong
Hungary
Italy
Liechtenstein
Luxembourg
| VoIP Providers
|
Configuration Guide
|
| IPnexia
|
-
|
Mexico
Netherlands
New Zealand
Portugal
Serbia
Slovakia
Slovenia
South Africa
Spain
Sweden
| VoIP Providers
|
Configuration Guide
|
| Tele2
|
-
|
| Telia
|
-
|
Switzerland
United Kingdom
United States of America
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 OpenScape Business V3R3.2
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).
External Links
Home 
A1 Telekom Austria SIP Trunk registrierend Konfiguration
Telefonica vivo SIP Trunk Configuration
