Difference between revisions of "Collaboration with VoIP Providers"

Revision as of 09:16, 29 April 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.

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

SIP trunking features

• One active SIP provider
• Up to 10 SIP user accounts
• SIP trunking with single numbers or 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 solve problems with network address translation between private and public IP adresses the following scenarios are supported
  • BizIP AD 20 with integrated SIP-aware Firewall connected directly to modem
  • STUN protocol enabled when BizIP is operated behind 3rd party router
  • Session boarder controller solves problem at Provider Network
• 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.

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.

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.

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.

d = default setting available from drop-down box in the WBM of HiPath BizIP
= 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	Type	HiPath BizIP	HiPath 2000	HiPath 3000	HiPath OpenOffice ME
inode	MSN

Canada

VoIP Providers	Type	HiPath BizIP	HiPath 2000	HiPath 3000	HiPath OpenOffice ME
thinktel

Denmark

VoIP Providers	Type	HiPath BizIP	HiPath 2000	HiPath 3000	HiPath OpenOffice ME
Viptel	MSN

France

VoIP Providers	Type	HiPath BizIP	HiPath 2000	HiPath 3000	HiPath OpenOffice ME
Plug & Tel

Germany

VoIP Providers	Type	HiPath BizIP	HiPath 2000	HiPath 3000	HiPath OpenOffice ME
1&1	MSN	d
AdvanceCall
Arcor	DID	d
bellshare
Broadnet	MSN DID	d
carpo
dus.net	MSN DID	d
freenet	MSN	d
freenetBusiness	DID	d
GMX	MSN	d
nikotel	MSN	d
PBX-network	MSN	d
PepPhone
Purtel	MSN	d
QSC	DID	d
SimplyConnect
sipgate	MSN	d
sipNetworks.de
STRATO	MSN	d
T-Online	MSN	d
T-Systems BAIP	DID
toplink	MSN DID	d
Vortel	MSN	d
XSip	DID	d

Greece

VoIP Providers	Type	HiPath BizIP	HiPath 2000	HiPath 3000	HiPath OpenOffice ME
FORTHnet
HellasOnLine
ViVodi

Hungary

VoIP Providers	Type	HiPath BizIP	HiPath 2000	HiPath 3000	HiPath OpenOffice ME
Megafone / Externet

Italy

VoIP Providers	Type	HiPath BizIP	HiPath 2000	HiPath 3000	HiPath OpenOffice ME
Multilink
BT Italia	DID
Welcome Italia	DID
Exsorsa	MSN

Latvia

VoIP Providers	Type	HiPath BizIP	HiPath 2000	HiPath 3000	HiPath OpenOffice ME
Lattelecom		d

The Netherlands

VoIP Providers	Type	HiPath BizIP	HiPath 2000	HiPath 3000	HiPath OpenOffice ME
I2Connect	MSN DID	d
Infopact	MSN DID	d
Priority Telecom	MSN DID	d

Portugal

VoIP Providers	Type	HiPath BizIP	HiPath 2000	HiPath 3000	HiPath OpenOffice ME
G9-Telecom
netcall
Portugal Telecom

Slovenia

VoIP Providers	Type	HiPath BizIP	HiPath 2000	HiPath 3000	HiPath OpenOffice ME
Detel	DID	d
UPC Telemach	DID	d
Amis	DID	d

Spain

VoIP Providers	Type	HiPath BizIP	HiPath 2000	HiPath 3000	HiPath OpenOffice ME
vozelia	MSN DID

United Kingdom

VoIP Providers	Type	HiPath BizIP	HiPath 2000	HiPath 3000	HiPath OpenOffice ME
sipgate (UK)	MSN	d
VoiceFlex	MSN DID	d
Gamma Telecom	MSN

United States of America

VoIP Providers	Type	HiPath BizIP	HiPath 2000	HiPath 3000	HiPath OpenOffice ME
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: 2008-04-25

Connectivity Test Information

The connectivity test is carried out with

• HiPath BizIP
• HiPath 2000
• HiPath 3000
• 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.

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.

External Links

• Siemens Enterprise Business Area (SEBA) (login required, Internet Explorer only)
•   VoIP Anbieter (Übersicht, Feature, Vergleiche)