Difference between revisions of "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.
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Revision as of 08:05, 9 December 2010
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!
.
Contents
- 1 Overview
- 2 General Information by platform
- 3 Tested VoIP Providers by Countries
- 3.1 International
- 3.2 Australia
- 3.3 Austria
- 3.4 Belgium
- 3.5 Brazil
- 3.6 Czech Republic
- 3.7 Chile
- 3.8 Finland
- 3.9 France
- 3.10 Germany
- 3.11 Italy
- 3.12 Mexico
- 3.13 The Netherlands
- 3.14 Portugal
- 3.15 Slovenia
- 3.16 South Africa
- 3.17 Spain
- 3.18 Sweden
- 3.19 Switzerland
- 3.20 United Kingdom
- 3.21 United States of America
- 4 Released SIP Providers in Detail
- 5 Frequently asked Questions
- 6 External Links
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 depends on available DSL bandwidth and used Codec, maximum number of simultaneous IP calls is 32
- automatic fallback to ISDN at SIP trunk failure
- Voice Codecs G.711 and G.729
- Fax over IP via T.38 (or G.711)
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.
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 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.
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.
VOIP provider test list and configuration last update: 2010-11-17
- 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
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.
OpenScape Office MX
SIP Trunking features
- All features, requirements and restrictions listed in overview are valid.
- Up to 32 simultaneous calls via SIP trunking (depending on available DSL bandwidth and used Codec)
- 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. (Web based mobility clients will solve this use case)
- FAX over IP with T.38 is supported for analog Fax devices but not for FAX communication with the OpenScape Office application, a TDM Trunk has to be used for that. (The feature is planned for V3).
- SIP Trunks are not released for use with the OpenScape Office Contact Center Application.
- A trunk prefix to select a SIP trunk as an alternate trunk can not be used from OpenScape Office Clients in V2 (myPortal, myAttendant, myAgent).
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.
= 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 MX |
|---|---|---|---|
| COLT | DID |
|
|
| Skype Connect | MSN |
|
|
- 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.
Our SME Platforms have been officially certified by Skype.
Australia
| VoIP Providers | Type | HiPath 3000 | OpenScape Office MX |
|---|---|---|---|
| Engin | MSN |
|
|
| Internode | MSN |
|
|
Austria
| VoIP Providers | Type | HiPath 3000 | OpenScape Office MX |
|---|---|---|---|
| UPC / inode | MSN |
|
- |
| WNT | DID |
|
- |
Belgium
| VoIP Providers | Type | HiPath 3000 | OpenScape Office MX |
|---|---|---|---|
| Belgacom | DID |
|
|
| Hexacom | MSN DID |
|
- |
| Telenet | MSN DID |
|
|
Brazil
| VoIP Providers | Type | HiPath 3000 | OpenScape Office MX |
|---|---|---|---|
| CTBC 1) | DID |
|
- |
| GVT 1) | DID |
|
|
Czech Republic
| VoIP Providers | Type | HiPath 3000 | OpenScape Office MX |
|---|---|---|---|
| Telefonica O2 | MSN DID |
|
|
Chile
| VoIP Providers | Type | HiPath 3000 | OpenScape Office MX |
|---|---|---|---|
| Entel NGN | MSN |
|
- |
Finland
| VoIP Providers | Type | HiPath 3000 | OpenScape Office MX |
|---|---|---|---|
| Elisa | DID |
|
- |
France
| VoIP Providers | Type | HiPath 3000 | OpenScape Office MX |
|---|---|---|---|
| OpenIP | DID | - |
|
| Optimitel | DID |
|
- |
Germany
| VoIP Providers | Type | HiPath 3000 | OpenScape Office MX |
|---|---|---|---|
| 1&1 | MSN |
|
- |
| Vodafone (Arcor) | VoIP Anlagenanschluss (DID) |
|
|
| freenet | MSN |
|
- |
| freenetBusiness | DID |
|
|
| GMX | MSN |
|
- |
| Purtel | MSN |
|
- |
| QSC | IPfonie extended (DID) |
|
|
| sipgate | sipgate basic (MSN) sipgate trunking (DID) |
|
|
| toplink | MSN DID |
|
|
| XSip | DID |
|
- |
Italy
| VoIP Providers | Type | HiPath 3000 | OpenScape Office MX |
|---|---|---|---|
| Infracom (ex Multilink) | MSN |
|
- |
Mexico
| VoIP Providers | Type | HiPath 3000 | OpenScape Office MX |
|---|---|---|---|
| Telefonica Mexico 1) | MSN | - |
|
The Netherlands
| VoIP Providers | Type | HiPath 3000 | OpenScape Office MX |
|---|---|---|---|
| I2Connect | MSN DID |
|
- |
| Infopact | MSN DID |
|
- |
| OneCentral | MSN DID |
|
|
| Priority Telecom | MSN |
|
|
| Tele2 | MSN DID |
- | 1)1)
|
1) For Openscape Office MX V2 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 MX |
|---|---|---|---|
| Portugal Telecom | DID |
|
|
Slovenia
| VoIP Providers | Type | HiPath 3000 | OpenScape Office MX |
|---|---|---|---|
| Amis | MSN |
|
|
| Detel | MSN |
|
|
| MegaM | MSN |
|
|
| tustelekom | DID |
|
|
| T-2 | DID | - |
|
| UPC Telemach | MSN |
|
|
South Africa
| VoIP Providers | Type | HiPath 3000 | OpenScape Office MX |
|---|---|---|---|
| Telfree | MSN DID |
- |
|
| Vodacom | MSN DID |
|
|
Spain
| VoIP Providers | Type | HiPath 3000 | OpenScape Office MX |
|---|---|---|---|
| vozelia | MSN DID |
|
- |
| Xtra Telecom | MSN |
|
|
Sweden
| VoIP Providers | Type | HiPath 3000 | OpenScape Office MX |
|---|---|---|---|
| Tele2 | DID |
| |
| Telia | MSN |
|
|
Switzerland
| VoIP Providers | Type | HiPath 3000 | OpenScape Office MX |
|---|---|---|---|
| Cablecom | DID |
|
|
| sipcall | MSN | - |
|
| SofiCall | MSN | - |
|
| vtx | DID |
|
|
United Kingdom
| VoIP Providers | Type | HiPath 3000 | OpenScape Office MX |
|---|---|---|---|
| Gamma | DID |
|
|
| Opal | DID |
|
|
| Voiceflex | DID |
|
|
United States of America
| VoIP Providers | Type | HiPath 3000 | OpenScape Office MX |
|---|---|---|---|
| Cbeyond | MSN |
|
- |
| Sotel | 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.
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: 2010-12-09
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
- 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)
