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!
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.
- Hold, Consultation, Toggle
- Transfer (attended, semi-attended)
- Ringing group, Call pickup
- 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 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 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.
- 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.
- 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.
VOIP provider data collection and test list - last update: 2016-08-12
--> Hint: Simplified and optimized test procedure from OSBiz V2.
- 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.
- Deliver the test results, including all Wireshark traces and final provider data to the VoIP Provider certification team.
- 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).
- 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.
OpenScape Business Configuration Guide for Internet Telephony:
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.
= has been successfully tested by development or customer
DID = accounts for PBX-trunking / Direct Inward Dialling
MSN = accounts for single number(s)
- 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
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.
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.
The following scenarios are covered within the connectivity test:
- Basic Calls from / to ITSP
- Feature Tests
- Consultation, transfer, toggle, conference
- Call Pickup / ringing group scenarios
- call forwarding scenarios
- CLIP / CLIR
- COLP / COLR
- 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