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Difference between revisions of "Collaboration with VoIP Providers"

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<br>'''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.
 
<br>'''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.
 
<br>'''2)''' No fixed servers (manual config needed), two profiles with different server addresses and identical accounts and MSN shall be used.
 
<br>'''2)''' No fixed servers (manual config needed), two profiles with different server addresses and identical accounts and MSN shall be used.
<br>'''3)''' ITSP does not support Fax T.38 Standard. <br>'''4)''' ITSP is certified for SIP-MEX functionality in OpenScape Business only.
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<br>'''4)''' ITSP is certified for SIP-MEX functionality.
 
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Revision as of 14:00, 15 August 2016

File:VoIP Provider SME.jpg
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!

Contents

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 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
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 http://wiki.unify.com/wiki/Features_and_Configuration_of_SIP_Devices

The following changes have been made starting from HiPath 3000 V9 OpenScape Office V3R2 and OpenScape Business V1:
  • 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 or OpenScape Business.

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 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. The ITSP MUST support T.38 if UC Suite fax via SIP trunk shall be used. (HiPath 3000 and OpenScape Office) 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 not available with HiPath 3000 and OpenScape Office. It 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.
  • 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. 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.
    zip.png  VOIP provider data collection and test list - last update: 2016-08-12

--> 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

New certifications will be based on OpenScape Business in future. 

  • 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.

Configuration

OpenScape Business Configuration Guide for Internet Telephony:

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

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

pdf.png  HowTo-CLIPnoScreeningEng- latest update: 2015-10-26

pdf.png  HowTo-CLIPnoScreeningDeu- latest update: 2015-11-04

pdf.png  Vodafone IP-ALA-R3 Germany- latest update: 2016-01-25

pdf.png  BCOM Netherlands- latest update: 2016-03-21

pdf.png  Deutsche Telekom DIPVD- latest update: 2016-06-03

pdf.png  KCOM UK ITSP SIP Trunk Configuration- latest update: 2016-06-03

pdf.png  X2COM Netherlands- latest update: 2016-06-03

HiPath 3000

Hint: OpenScape Business is the successor of HiPath 3000. Since HiPath 3000 is discontinued, there are no further certification activities.

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:

pdf.png  HiPath 3000 Config Guide ITSP

See also:

How to configure CLIP sent via SIP trunk / ITSP

OpenScape Office LX/MX

Valid for OpenScape Office V3 LX / MX.

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 UC Suite application. Fax via G.711 is not supported with the UC Suite application. The fax functionality UC Suite only supports the transmission T.38 standard. Prerequisite for the correct fax transmission is a connection via an ISDN gateway or an ITSP that supports T.38.
  • 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:

See also:


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.

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 OpenScape Business
COLT DID tick.gif
Verizon   MSN / DID tick.gif

pdf.png  AppNote OpenScape_Business_V1 with_Verizon_IP_Trunks ML 10052013 V05

Skype Connect   MSN tick.gif
  • 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.

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

Australia

VoIP Providers Type OpenScape Business
Engin MSN tick.gif
Internode MSN tick.gif
Commander Primus MSN / DID tick.gif
Telstra MSN / DID tick.gif

Austria

VoIP Providers Type OpenScape Business
UPC / inode   MSN tick.gif
IP Austria / my Tweak   DID tick.gif
/Kabelplus Business Phone   DID tick.gif
/ LinzAG City Voice   DID tick.gif
/ Neotel   DID tick.gif
/ SalzburgAG CableLink   DID tick.gif
/ Tele2   DID tick.gif

Belgium

VoIP Providers Type OpenScape Business
Belgacom   DID tick.gif
Belgacom IMS   MSN / DID tick.gif
Destiny MSN / DID tick.gif
Hexacom   MSN / DID tick.gif
Telenet MSN / DID tick.gif
WIN   MSN / DID tick.gif

Brazil

VoIP Providers Type OpenScape Business
CTBC   1) DID tick.gif
GVT   1) DID tick.gif
Voitel MSN tick.gif

Canada

VoIP Providers Type OpenScape Business
BabyTEL   DID tick.gif

Czech Republic

VoIP Providers Type OpenScape Business
GTS DID tick.gif
T-mobile MSN / DID tick.gif
O2 MSN / DID tick.gif

Chile

VoIP Providers Type OpenScape Business
Entel NGN MSN tick.gif

Denmark

VoIP Providers Type OpenScape Business
Uni-tel A/S MSN / DID tick.gif
Global Connect DID tick.gif

Finland

VoIP Providers Type OpenScape Business
Elisa   DID tick.gif
Länsilinkki MSN tick.gif
Telia-Sonera DID tick.gif
TDC MSN / DID tick.gif

France

VoIP Providers Type OpenScape Business
Acropolis Telecom   DID tick.gif
MyStream   MSN / DID tick.gif
OpenIP   DID tick.gif
Paritel Operateur MSN / DID tick.gif
Completel DID tick.gif
Hexatel DID tick.gif
Sewan Communications MSN / DID tick.gif
Bouygues MSN / DID tick.gif

Germany

VoIP Providers Type OpenScape Business
1&1   MSN tick.gif
Vodafone (formerly Arcor)   VoIP Anlagenanschluss (DID) tick.gif
Vodafone   Vodafone Anlagenanschluss R3 (DID) tick.gif
ennit AG(formerly tng) DID / MSN tick.gif
GMX (now 1&1)   MSN tick.gif
Purtel   MSN tick.gif
QSC IPfonie extended (DID) tick.gif
QSC IPfonie extended Connect (DID) tick.gif
sipgate   sipgate basic (MSN)
sipgate trunking (DID)
tick.gif
toplink   MSN / DID tick.gif
Deutsche Telekom   Call&Surf IP (MSN) tick.gif
ecotel sipTrunk DDI DID tick.gif
ecotel sipTrunk Connect 1.0 DID tick.gif
M-Net   DID tick.gif
MK Netzdienste   DID tick.gif
Peoplefone AG   DID tick.gif
Equada   MSN / DID tick.gif
HFO   MSN / DID tick.gif

Italy

VoIP Providers Type OpenScape Business
Infracom (ex Multilink)   MSN tick.gif
TWT MSN / DID tick.gif
Viatek MSN / DID tick.gif

Mexico

VoIP Providers Type OpenScape Business
Telefonica Mexico   1) MSN tick.gif

The Netherlands

VoIP Providers Type OpenScape Business
Infopact   MSN / DID tick.gif
OneCentral   4) MSN / DID tick.gif
OneXS MSN / DID tick.gif
Priority Telecom   MSN tick.gif
RoutIT DID tick.gif
Speakup   MSN / DID tick.gif
Tele2 MSN / DID tick.gif
Vodafone NL MSN / DID tick.gif
Voiceworks   DID tick.gif
Ziggo   DID tick.gif
T-Mobile NL MSN / DID tick.gif
Motto Communications MSN / DID tick.gif
BCOM DID tick.gif
X2COM DID tick.gif

New Zealand

VoIP Providers Type OpenScape Business
TelstraClear   MSN / DID tick.gif
ORCON MSN / DID tick.gif

Portugal

VoIP Providers Type OpenScape Business
Portugal Telecom DID tick.gif

Serbia

VoIP Providers Type OpenScape Business
Telenor   DID tick.gif

Slovakia

VoIP Providers Type OpenScape Business
Slovak Telekom   DID tick.gif

Slovenia

VoIP Providers Type OpenScape Business
Amis   MSN tick.gif
Detel   MSN tick.gif
MegaM   MSN tick.gif
tustelekom   DID tick.gif
T-2   DID tick.gif
Telemach   MSN tick.gif

South Africa

VoIP Providers Type OpenScape Business
Nashua NCN MSN / DID tick.gif
Telfree   MSN / DID tick.gif
Vodacom MSN / DID tick.gif

Spain

VoIP Providers Type OpenScape Business
LCR (Least Cost Routing) MSN / DID tick.gif
vozelia   MSN / DID tick.gif
Xtra Telecom   MSN tick.gif
Telecable DID tick.gif

Sweden

VoIP Providers Type OpenScape Business
Tele2   4) DID tick.gif
Telia   MSN tick.gif

Switzerland

VoIP Providers Type OpenScape Business
Cablecom DID tick.gif
efon MSN / DID tick.gif
sipcall MSN tick.gif
SofiCall   MSN tick.gif
Sunrise MSN / DID tick.gif
Swisscom BCON DID tick.gif
vtx   DID tick.gif
Telco Pack SA   DID tick.gif
Peoplefone AG   DID tick.gif
Swisscom Smart Business Communication   DID tick.gif
Swisscom VoIP Gate   DID tick.gif

United Kingdom

VoIP Providers Type OpenScape Business
BT IPVS MSN / DID tick.gif
Gamma   DID tick.gif
HIPCOM MSN / DID tick.gif
Opal   DID tick.gif
Voiceflex   DID tick.gif
VoIP Ltd i-Line   MSN / DID tick.gif
tIPicall   DID tick.gif
TalkTalk   N.A. tick.gif
KCOM   DID tick.gif

United States of America

VoIP Providers Type OpenScape Business
AT&T DID

pdf.png  OSO_ATT SipTrk Configuration Note 07092013 V1

tick.gif
Cbeyond   MSN tick.gif
CenturyLink   2) DID tick.gif
Sotel   MSN / DID tick.gif
Windstream  

(Broadsoft platform only)

MSN / DID tick.gif
XO   1) MSN / DID tick.gif
Intermedia   DID 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.
2) No fixed servers (manual config needed), two profiles with different server addresses and identical accounts and MSN shall be used.
4) ITSP is certified for SIP-MEX functionality.

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.

pdf.png  Overview about Released SIP Providers Latest update: 2016-06-17

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

External Links