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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The connectivity test is carried out with | The connectivity test is carried out with | ||
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* [[HiPath 2000]] | * [[HiPath 2000]] | ||
* [[HiPath 3000]] | * [[HiPath 3000]] | ||
+ | * [[HiPath OpenOffice EE]] | ||
* [[HiPath OpenOffice ME]] | * [[HiPath OpenOffice ME]] | ||
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. | 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. |
Revision as of 09:15, 1 December 2008
Collaboration with VoIP Providers via SIP Trunking is an important topic in the todays world of Voice/Data Communication over the Internet. The following information will give an overview of the supported VoIP providers in the different countries as well as additional considerations and hints to the connected platforms.
Contents
Overview
HiPath Platforms have been tested with a large number of Internet telephony providers that support SIP. At the moment, there are differences in the range of features supported, reachable networks, and service quality offered by the various providers. Occasionally configuration changes may occur that require adjustments to be made to the HiPath Platforms.
General Information by platform
For discontinued platforms see Collaboration with VoIP Providers (discontinued platforms).
HiPath OpenOffice EE / HiPath 2000
SIP trunking features
- 4 active VoIP Providers
- Up to 30 SIP user accounts
- E2E payload enables an increase of simultaneous calls to the Internet Telephony Service Provider. The limitation of 4 simultaneous calls does not apply any more
- Fax over IP via T.38 or G.711 (T.38 towards ITSPs is not supported)
General requirements
- LAN with 10/100/1000 MBit/s.
- 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.
- Not more than 20 ms jitter.
- Quality of Service (QoS) support - IEEE 802.p, DiffServ (RFC 2474) or ToS (RFC 791).
- Not more than 40% network load.
LANs that are linked over WANs and share the same VoIP functionality must meet the following minimum requirements:
- Each LAN must be connected to the Internet via a WAN with a fixed IP address.
- The upload and download bandwidth required for calls must always be available.
- The maximum number of simultaneous VoIP connections via WAN is dependent on factors, such as the codecs used.
- In addition to DSL, radio and laserlink are also WAN connections.
- HiPath 2000 does not have an integrated modem and therefore requires an external modem, such as a DSL modem.
- If HiPath 2000 has set the NAT flag when connected to an interface (WAN or PPP/ISDN), HiPath 2000 and the local LAN are protected.
Configuration
See HiPath OpenOffice EE V1 Configuration Guide for Internet Service Providers:
SIP Provider dependent restrictions:
- 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 (110, 112) are not supported since the local network cannot be clearly assigned when exiting into the public network. These connections must be implemented via S0 access and are generally provided in advance by the setup assistant.
- Calling line identification
The CLIP and CLIR features are implemented differently by the various providers. Limitations in the calling line identification may occur. - For connections that exit to the fixed/wireless network or come in from the fixed/wireless network:
- Provider-internal connections (destination is busy/unregistered) Internal connections to a busy or unregistered user are generally correctly signaled.
- Calls to other providers
These calls are technically possible but the individual providers adapt the routing functions of their network to each particular business model. More information is available with the respective SIP provider.
- Call data evaluation
The SIP protocol records individual call data against time. The data can be evaluated using an external application such as Teledata Office. - DTMF transmission over SIP:
- DTMF Inband Signalling happens by default.
- Might be necessary to disable "DTMF transmission according to RFC 2833" when no DTMF tones passes via the DSL line.
- Might be necessary to disable "DTMF Outband Signalling" at the same time and in some specific cases too.
- FAX via SIP Provider: FAX transmissions are routed via ISDN by default: HiPath 2000 V1.0 permits FAX transmissions to be routed via SIP provider but some issues should be considered:
- FAX transmission is only possible with Codec G.711.
- Automatic switchover from G.723 or G.729 to G.711 is not possible.
- Most SIP Providers do not support T.38 FAX transmission over SIP since they haven't it yet fully implemented.
- Packets losses may occur and be too high using Codec G.711 for FAX transmission over SIP via DSL connection due to higher bandwidth requirements of this codes compared to the others: FAX transmissions could be fully or partially incomplete.
HiPath 3000
HiPath 3000 V7 with HG 1500 can be used for SIP trunking in the same way as HiPath 2000 V2.
An external router is always necessary.
HiPath OpenOffice ME
SIP trunking features
- 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) or Direct Inward Dialing (DID)
- number of simultaneous VoIP connections is only limited by DSL bandwidth
- automatic fallback to ISDN at SIP trunk failure
- G.711 and G.729 Codecs are supported by the OpenOffice appliance
- Fax over IP with T.38 planned for minor release 4
General requirements
- Intermediate routers shall support quality of service and bandwidth control mechanisms if the WAN link is used for voice and data simultaneously.
SIP Provider dependent restrictions
- to be provided -
Tested VoIP providers by countries
The main SIP trunking functions of the HiPath platforms have been tested with the following Internet telephony providers. For detailed information see 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 2000 | HiPath 3000 | HiPath OpenOffice EE | HiPath OpenOffice ME |
---|---|---|---|---|---|
COLT | DID |
Please check provider home page for supported countries.
Australia
VoIP Providers | Type | HiPath 2000 | HiPath 3000 | HiPath OpenOffice EE | HiPath OpenOffice ME |
---|---|---|---|---|---|
engin | MSN DID |
|
|
Austria
VoIP Providers | Type | HiPath 2000 | HiPath 3000 | HiPath OpenOffice EE | HiPath OpenOffice ME |
---|---|---|---|---|---|
inode | MSN | ||||
WNT | DID |
Canada
VoIP Providers | Type | HiPath 2000 | HiPath 3000 | HiPath OpenOffice EE | HiPath OpenOffice ME |
---|---|---|---|---|---|
thinktel |
Germany
VoIP Providers | Type | HiPath 2000 | HiPath 3000 | HiPath OpenOffice EE | HiPath OpenOffice ME |
---|---|---|---|---|---|
1&1 | MSN | ||||
Arcor | DID | ||||
freenet | MSN | ||||
freenetBusiness | DID | ||||
GMX | MSN | ||||
Purtel | MSN | ||||
QSC | DID | ||||
sipgate | MSN DID |
[1] | |||
T-Online | MSN | ||||
T-Systems BAIP | DID | ||||
toplink | MSN DID |
|
|
|
|
XSip | DID |
- ↑ For availability of DDI account by sipgate, please refer to http://www.sipgate.de
Greece
VoIP Providers | Type | HiPath 2000 | HiPath 3000 | HiPath OpenOffice EE | HiPath OpenOffice ME |
---|---|---|---|---|---|
FORTHnet | |||||
HellasOnLine | |||||
ViVodi |
Italy
VoIP Providers | Type | HiPath 2000 | HiPath 3000 | HiPath OpenOffice EE | HiPath OpenOffice ME |
---|---|---|---|---|---|
Infracom (ex Multilink) | |||||
BT Italia | DID | ||||
Welcome Italia | DID |
The Netherlands
VoIP Providers | Type | HiPath 2000 | HiPath 3000 | HiPath OpenOffice EE | HiPath OpenOffice ME |
---|---|---|---|---|---|
I2Connect | MSN DID |
|
|||
Infopact | MSN DID |
|
|||
Priority Telecom | MSN DID |
Portugal
VoIP Providers | Type | HiPath 2000 | HiPath 3000 | HiPath OpenOffice EE | HiPath OpenOffice ME |
---|---|---|---|---|---|
G9-Telecom | |||||
netcall |
Slovenia
VoIP Providers | Type | HiPath 2000 | HiPath 3000 | HiPath OpenOffice EE | HiPath OpenOffice ME |
---|---|---|---|---|---|
Amis | MSN DID |
||||
Detel | MSN DID |
||||
tustelekom | DID | ||||
UPC Telemach | MSN DID |
Spain
VoIP Providers | Type | HiPath 2000 | HiPath 3000 | HiPath OpenOffice EE | HiPath OpenOffice ME |
---|---|---|---|---|---|
vozelia | MSN DID |
|
|
|
|
Xtra Telecom | MSN DID |
Sweden
VoIP Providers | Type | HiPath 2000 | HiPath 3000 | HiPath OpenOffice EE | HiPath OpenOffice ME |
---|---|---|---|---|---|
Telia | MSN |
Switzerland
VoIP Providers | Type | HiPath 2000 | HiPath 3000 | HiPath OpenOffice EE | HiPath OpenOffice ME |
---|---|---|---|---|---|
sipcall | MSN | ||||
vtx | DID |
United Kingdom
VoIP provider release for Voiceflex and others is pending in UK.
United States of America
VoIP Providers | Type | HiPath 2000 | HiPath 3000 | HiPath OpenOffice EE | HiPath OpenOffice ME |
---|---|---|---|---|---|
Cbeyond | |||||
Sotel | MSN DID |
|
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: 2008-10-07
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)