Difference between revisions of "Collaboration with VoIP Providers"

Revision as of 23:18, 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.

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.

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. For detailed information see the document Overview over released SIP providers below.

def = 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

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		def
AdvanceCall
Arcor	DID
bellshare
Broadnet	MSN DID	def
carpo
dus.net	MSN DID	def
freenet	MSN	def
freenetBusiness	DID	def
GMX		def
nikotel		def
PBX-network
PepPhone
Purtel
QSC	DID	def
SimplyConnect
sipgate		def
sipNetworks.de
STRATO		def
T-Online		def
toplink	MSN DID	def
Vortel
XSip	DID

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

The Netherlands

VoIP Providers	Type	HiPath BizIP	HiPath 2000	HiPath 3000	HiPath OpenOffice ME
I2Connect		def
Infopact		def
Priority Telecom		def

Portugal

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

Slovenia

VoIP Providers	Type	HiPath BizIP	HiPath 2000	HiPath 3000	HiPath OpenOffice ME
Detel	DID	def
Amis

United Kingdom

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

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: 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
  • Consultation, transfer, toggle, conference
  • ringing group scenarios
  • call forwarding scenarios
  • DTMF
  • CLIP / CLIR
  • COLP / COLR
  • FAX

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)