Collaboration with VoIP Providers



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!'''

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)
 * 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
 * large number of simultaneous VoIP connections depending on 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)

Enhanced SIP attack protection has been implemented with HiPath 3000 V9 and OpenScape Office V3R2. For diagnostics and SIP provider configuration hints see How to enable blocked SIP Providers.

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. The tested SIP Stack is used in the current system versions Hipath 3000 V8 and OpenScape Office V3 as well.

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. One common STUN setting must be used for all providers in a system.

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.
 * The ITSP has to support outband DTMF (RFC2833) for using the OpenScape Office Autoattendant via SIP trunk.
 * CLIP no screening is not available with our platforms for SIP trunks. Please check with your ITSP, if this can be accomplished at the provider side.

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 [mailto:franz.kneissl@siemens-enterprise.com product management].
 * 2) Get the IP and SIP settings for the specific provider. Send the completed document "VoIP Provider Data Collection" to the [mailto:anastasios.ginis@siemens-enterprise.com;karl-werner.weigt@siemens-enterprise.com VoIP Provider certification team]. This is essential for correct PBX setup. The tables in chapter 2 have to be filled in. - last update: 2012-04-12
 * 3) Perform all tests in the VoIP provider test list, according to the instructions included. Remote support is available by the [mailto:anastasios.ginis@siemens-enterprise.com;karl-werner.weigt@siemens-enterprise.com VoIP Provider certification team]. Leading switch should be the OpenScape Office MX. After that, a reduced delta test list can be used for HiPath 3000. - last update: 2012-04-12
 * 4) Deliver the test results, including all Wireshark traces and final provider data to the [mailto:anastasios.ginis@siemens-enterprise.com;karl-werner.weigt@siemens-enterprise.com VoIP Provider cerification 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).

HiPath 3000
Valid for HiPath 3000 ≥ V7 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.
 * No FAX via ITSP from/to OpenScape Office HX client applications. A TDM trunk has to be used.
 * A trunk prefix to select a SIP trunk as an alternate trunk can be used from the OpenScape Office HX client's dial box but not from client's directories and settings (myPortal, myAttendant, myAgent).
 * SIP trunking is released for use with the OpenScape Office HX Contact Center application starting from HiPath 3000 V9.

Configuration

HiPath 3000 Configuration Guide for Internet Telephony:



See also:
 * How to configure CLIP sent via SIP trunk / ITSP

OpenScape Office LX/MX
Valid for OpenScape Office V2 (MX) and 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 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. 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. No Fax over IP is available for the OpenScape Office V2 application, a TDM Trunk has to be used.
 * 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.
 * SIP trunking is released for use with the OpenScape Office Contact Center application starting from OpenScape Office V3R2.

Configuration

OpenScape Office Configuration Guide for Internet Telephony:



See also:
 * How to enter SIP Provider Account Data
 * How to configure CLIP sent via SIP trunk / 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.

Yes = has been successfully tested by development or customer DID = accounts for PBX-trunking / Direct Inward Dialling MSN = accounts for single number(s)

International

 * 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.

United States of America
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.

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.


 * Latest update: 2012-04-25

Connectivity Test Information

The connectivity test is carried out with


 * HiPath 3000
 * OpenScape Office MX/LX

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
 * 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

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