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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* Up to 8 simultaneous VoIP connections, depending on DSL bandwidth | * Up to 8 simultaneous VoIP connections, depending on DSL bandwidth | ||
* Parallel ISDN and VoIP connections | * Parallel ISDN and VoIP connections | ||
| − | * Fax via ISDN or SIP terminal adapter | + | * Fax via ISDN or SIP terminal adapter for Fax T.38 |
'''General requirements''' | '''General requirements''' | ||
| − | * HiPath BizIP should be connected directly to the xDSL modem through WAN interface. In | + | * HiPath BizIP should be connected directly to the xDSL modem through WAN interface. In special cases [[HiPath BizIP Operation behind existing Routers|operation behind existing routers]] is possible. |
| − | * To use the built-in voice mail, the provider must support [[Codec#G.729|G.729]] voice compression. | + | * To use the built-in voice mail, the provider must support [[Codec#G.729|G.729]] voice compression. At test time e.g. AdvanceCall, PepPhone, Purtel and SimplyConnect did not support G.729. |
* For remote access to the answering machine, the provider must support outband tone dialing transmission (DTMF) in accordance with RFC 2833. | * For remote access to the answering machine, the provider must support outband tone dialing transmission (DTMF) in accordance with RFC 2833. | ||
Revision as of 22:24, 17 November 2007
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
HiPath BizIP
- One active SIP provider
- Up to 10 SIP user accounts
- SIP Direct Inward Dialling
- Up to 8 simultaneous VoIP connections, depending on DSL bandwidth
- Parallel ISDN and VoIP connections
- Fax via ISDN or SIP terminal adapter for Fax T.38
General requirements
- HiPath BizIP should be connected directly to the xDSL modem through WAN interface. In special cases operation behind existing routers is possible.
- To use the built-in voice mail, the provider must support G.729 voice compression. At test time e.g. AdvanceCall, PepPhone, Purtel and SimplyConnect did not support G.729.
- For remote access to the answering machine, the provider must support outband tone dialing transmission (DTMF) in accordance with RFC 2833.
SIP Provider dependent restrictions
- Special call numbers and emergency calls are not supported by all providers. Emergency calls are routed automatically through ISDN by HiPath BizIP. Special call numbers can be defined to be routed through ISDN by Web based Management.
- Calling line identification: The CLIP and CLIR features are implemented differently by the various providers. Limitations in the calling line identification may occur. For some providers CLIP/CLIR has to be configured in the Providers' Web Portal.
- Some Providers do not support streaming of Musik on Hold (MOH) to a device in call hold (e.g. sipgate).
- Call Forwarding external - external is currently not possible with some providers because they do not support direct payload (e.g. T-Online, QSC, VoiceFlex).
For minor limitations or configuration details please see also the release note at ftp://bizip.siemens.com/HiPath_BizIP1.0.
HiPath 2000
The HiPath 2000 V1.0 introduced support of ITSPs (Internet Telephony Service Provider) supporting SIP Protocol over DSL Telephony since the release of SMR 6 (HE620Y.06.439) software version enhancing features, limits and capabilities with each new software version released.
HiPath 2000 V1.0 DSL Telephony features (Hints and restrictions):
- starting from HiPath 2000 V1.0 R9.5.0 (HE620Y.09.648):
- STUN support is provided and the system now connects to external customers' Router performing NAT (Routing devices that perform Symmetrical NAT are not supported).
- up to 4 active SIP Providers can be active at the same time on the system.
- the complete DID range (ITSP providers dependant) is available for the SIP Clients.
- DSL telephony can be operated in parallel with CorNet IP Trunking with no restrictions.
- up to HiPath 2000 V1.0 R7.0.0 (HE620Y.07.554):
- DSL Routing (NAT) was performed by the system connected to external xDSL Modem or xDSL Router with routing (NAT) section disabled.
- only one SIP Provider can be active on the system.
- up to 30 SIP Clients (MSN) can be configured at the same time on the system.
- DSL telephony can't be configured in parallel with CorNet IP Trunking.
- A maximum of 4 simultaneous connections to the SIP provider are supported per IP Phone (xDSL Bandwidth should be considered to prevent quality losses or service disruptions).
- Both S0 connections and SIP connections are supported in parallel via the Internet.
|
Networking scenarios with several HiPath 2000 or HiPath 2000 with HiPath 3000/HiPath 4000/ HiPath 5000, SMG involved use S0 trunk access only. |
System-related connection conditions
HiPath 2000 V1.0 should be connected directly to the xDSL modem through WAN interface (with custom PPPoE or predefined SIP Provider settings).
Since R9.5.0, STUN is now supported allowing the HiPath 2000 to be operated behind 3rd party Router (NAT)/Firewall.
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.
Tested VoIP providers by countries
HiPath platforms main functions have been tested with the following Internet telephony providers that support SIP.
default = general access data is already available as default setting in the WBM
= has been successfully tested by development or customer
DID = accounts for PBX-trunking / Direct Inward Dialling
MSN = accounts for single number(s)
Austria
| VoIP Providers | HiPath BizIP | HiPath 2000 | HiPath 3000 |
|---|---|---|---|
| inode |
|
|
Canada
| VoIP Providers | HiPath BizIP | HiPath 2000 | HiPath 3000 |
|---|---|---|---|
| thinktel |
|
France
| VoIP Providers | HiPath BizIP | HiPath 2000 | HiPath 3000 |
|---|---|---|---|
| Plug & Tel |
|
Germany
| VoIP Providers | HiPath BizIP | HiPath 2000 | HiPath 3000 |
|---|---|---|---|
| 1&1 | default | default | default |
| AdvanceCall | [1]
|
||
| Arcor | (DID)
|
(DID)
|
(DID)
|
| bellshare |
|
||
| Broadnet | default (MSN+DID) | ||
| carpo |
|
||
| dus.net | default (MSN+DID) | ||
| freenet | default | default | default |
| freenetBusiness | default (DID) | DID | DID |
| GMX | default |
|
|
| nikotel | default | ||
| PBX-network |
|
||
| PepPhone | [1]
|
||
| Purtel | [1]
|
|
|
| QSC | default (DID) | default (DID) | default (DID) |
| SimplyConnect | [1]
|
||
| sipgate | default | default | default |
| sipNetworks.de |
|
||
| STRATO | default | ||
| T-Online | default | default | default |
| toplink | default (DID) | default (MSN+DID) | default (MSN+DID) |
| Vortel |
|
||
| XSip | (DID)
|
(DID)
|
(DID)
|
Greece
| VoIP Providers | HiPath BizIP | HiPath 2000 | HiPath 3000 |
|---|---|---|---|
| FORTHnet |
|
||
| HellasOnLine |
|
||
| ViVodi |
|
Hungary
| VoIP Providers | HiPath BizIP | HiPath 2000 | HiPath 3000 |
|---|---|---|---|
| Megafone / Externet |
|
Italy
| VoIP Providers | HiPath BizIP | HiPath 2000 | HiPath 3000 |
|---|---|---|---|
| Multilink |
|
| |
| BT Italia | (DID)
|
||
| Welcome Italia | (DID)
|
||
| Exsorsa |
|
The Netherlands
| VoIP Providers | HiPath BizIP | HiPath 2000 | HiPath 3000 |
|---|---|---|---|
| I2Connect | default | ||
| Infopact | default | ||
| Priority Telecom | default |
Portugal
| VoIP Providers | HiPath BizIP | HiPath 2000 | HiPath 3000 |
|---|---|---|---|
| netcall |
|
| |
| Portugal Telecom |
|
Slovenia
| VoIP Providers | HiPath BizIP | HiPath 2000 | HiPath 3000 |
|---|---|---|---|
| Detel | default (DID) | ||
| Amis |
|
United Kingdom
| VoIP Providers | HiPath BizIP | HiPath 2000 | HiPath 3000 |
|---|---|---|---|
| sipgate (UK) | default |
|
|
| VoiceFlex | default (MSN+DID) |
|
|
| Gamma Telecom |
|
United States of America
| VoIP Providers | HiPath BizIP | HiPath 2000 | HiPath 3000 |
|---|---|---|---|
| Cbeyond | - |
|
|
Tests for the HiPath 2000 and HiPath 3000 are currently going to be conducted with Verizon.
Released SIP-Providers
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: 2007-11-08
Connectivity Test Information
The connectivity test is treated with HiPath platform
- BizIP
- HiPath 2000
- HiPath 3000
- HiPath OpenOffice
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
- Special Tests
- emergency calls
- service calls
- long duration calls
- Feature Tests
The connectivity tests are performed using the following endpoints creatively:
- 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 testcase.
External Links
- Siemens Enterprise Business Area (SEBA) (login required, Internet Explorer only)
-
VoIP Anbieter (Übersicht, Feature, Vergleiche)
