HiPath 3000/5000

Technical Bulletin No. 95

ITSP Configuration on HiPath 3000
Author: Alan G Bosworth & Paul Spencer

1 Introduction

The ability to configure SIP trunks for connection to an Internet Telephony Service Provider (ITSP) is
released from HiPath 3000 V6.0.

This document does not attempt to detail all the ITSP features available in the HiPath 3000, but is
produced only as an aide memoir for ITSP trunk configuration.

Before deciding to deploy an ITSP on-site please consult the ‘SIP Trunking Best Practice Guide’ for
SME platforms. This document will be available on the HiPath 3000 Engineers’ Toolkit DVD-ROM from
Issue 7 and can be downloaded from eTAC (ID 15294), follow the link below:

ITSP / SIP Trunking Best Practice Guide

NOTE #1 Up to four separate ITSPs may be configured for use at a time.

2 Prerequisites

 HiPath 3000 V6.0 R9.2.0 (Binder 09.644) or later

 HiPath HG1500 V6.0 R7.1.0 (Binder 67.003) or later

OR

STMI2 V6.0 R7.1.0 (Binder T7.003) or later

 The HiPath HG1500/STMI2 MUST have access to the Internet via the customers’ LAN.

 An account must have been setup with an ITSP certified in the UK.

The following ITSP has been certified in the UK:

VoiceFlex (http://www.voiceflex.com)

Make sure you have the following information from your ITSP before continuing:

 SIP gateway IP Address / DNS Host Name and Port

 User Name
 Password
 External numbers associated with this account

NOTE #2 IMPORTANT PLEASE READ Siemens Enterprise Communications Limited will only provide
support for certified ITSPs via eTAC and TAC. If you choose to use an uncertified ITSP it is unlikely to
function correctly in all scenarios even if the account appears to activate and register with the ITSP.
Siemens Enterprise Communications Limited will be unable to provide support in this case. If you
require a particular ITSP to be certified please contact your Siemens Account Manager.

3 Configuration

For the purpose of the example we will consider the configuration required on a HiPath 3000 V8 system,
any differences required for HiPath 3000 V6.0 / V7 will be pointed out in the text. Also HiPath 3000
Manager E V8 or later should be used, otherwise the screenshots displayed below may not match. The
HiPath 3000 configuration MUST be carried out BEFORE the HiPath HG1500/STMI2 configuration.

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3.1 HiPath 3000 Configuration

Configuration is carried out using HiPath 3000 Manager E; download the latest KDS from the system
before continuing. It is assumed that basic HiPath 3000 configuration has already been carried out.

1. Under the IP Trunks tab enter the Number of IP trunks required and select an unused ‘SIP
Provider’ from the drop-down list (e.g. SIP Provider 1) and click on the Add button. The trunks will
automatically be added to a trunk group matching the ‘SIP Provider’ chosen (e.g. ‘Trk Grp12’). Also
check the Enable gateway resources flag (if not already checked). Click on the Apply button.

NOTE #3 Four SIP providers can be configured on HiPath 3000 (SIP Provider 1 to SIP Provider 4)
and these are hard-wired to trunk groups 12 to 15 respectively.

2. In the Lines / networking object under the Routes tab select trunk group 12. Change the
route Name to identify the ITSP used (e.g. VoiceFlex). On HiPath 3000 V8 the LCR Digit
transmission method is assigned per trunk group and will automatically change to ‘en-bloc
sending’. This MUST be changed manually on HiPath 3000 V6.0 / V7 systems (step 4).

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NOTE #4 From HiPath 3000 V8 the Always use DSP flag is disabled in default as shown in the
screenshot above, this allows the direct payload feature to be used. The status of this flag MUST
match the configuration in the HiPath HG1500/STMI2 card. See section 4 for more information.

3. Under the Routing Parameters tab make sure all flags and parameters are set as per the
screenshot below.

4. In the Least cost routing object under the Flags and COS tab select the Activate LCR flag (if
not already enabled). On HiPath 3000 V6.0 / V7 systems change the Digit transmission method to
‘en-bloc sending’. On HiPath 3000 V8 systems this is configured automatically per trunk group (see
step 2).

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5. Select the Dial plan tab and add appropriate Dialed digits to allow calls to route via the ITSP as
per the customers’ requirements (e.g. ‘9CZ’). Link the Dialed digits to a free Route table (e.g. 1).
Add the ITSP Route (e.g. ‘VoiceFlex’) as the first entry in the Route table; it is also possible to add
PSTN trunks as the second entry in the route table (e.g. ‘BT’). In this case calls will overflow to the
PSTN if the ITSP fails or if all ITSP trunks are busy, a Warning (audible, visual or both) can be
provided when this occurs (e.g. ‘display + tone’). Make sure that the Dial rule is set to Main
network supplier with an appropriate Dial rule format (e.g. ‘E2A’). Certain types of call cannot be
routed over an ITSP e.g. emergency calls, Fax and Modem traffic. Check with your ITSP for any
other restrictions. Make sure that these calls route via the PSTN (e.g. ‘9C999’ and ‘9C112’ entries).
Also, make sure that any entries for emergency calls have the Emergency parameter set to ‘yes’.

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6. Upload the KDS to the system.

3.2 HiPath HG1500/STMI2 Configuration

Configuration is carried out using WBM. It is assumed that basic HiPath HG1500/STMI2 configuration
has already been carried out.

1. After adding the first SIP provider using HiPath 3000 Manager E the HiPath HG1500 / STMI2
card MUST be restarted by clicking on the Reset icon at the bottom of the WBM page.

2. Navigate to Explorers > Routing > IP Routing and click + to expand the IP Routing tree,
right-click Default Router and select Edit Default Router. Change Default Routing via to
‘LAN’ using the drop-down list and enter the IP Address of Default Router (e.g.
192.168.12.254) on the customers’ LAN. This may be the IP address of the customers ADSL
router. Click on the Apply button.

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3. Right-click DNS Server and select Edit DNS Settings. Enter the IP address of DNS Server;
this can be a DNS server on the customers’ LAN (if available), the ADSL router or an ISP DNS
server. Click on the Apply button.

4. Go to the Explorers > Voice Gateway screen and click + to expand the Internet Telephony
Service Provider tree.

ITSP Profile Available

If your ITSP is listed in the tree right-click the Internet Telephony Service Provider name
(e.g. Voiceflex) and check if the STUN server IP address has been entered in the Provider
STUN IP Address box, if not this MUST be obtained from the ITSP and entered before
continuing. As an example, the STUN server IP address for VoiceFlex MUST be entered
manually and is shown in the screenshot below. Click on the Apply button after making any
changes.

ITSP Profile Unavailable

If your ITSP is not listed in the tree it must be manually configured (However, see NOTE #2),
right-click Internet Telephony Service Provider and select Add Internet Telephony Service
Provider. Add the ITSP name to the Provider Name box and configure using the details
provided by the ITSP. The STUN server IP address MUST be entered in the Provider STUN
IP Address box. For illustration purposes the settings for VoiceFlex are shown in the
screenshot below although there is a preconfigured profile already available. Click on the
Apply button.

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5. Right-click the Internet Telephony Service Provider name (e.g. Voiceflex) and select Add
Internet Telephony Station. Enter the ITSP account user name as both the Internet
Telephony Station and the Authorization name and enter the ITSP account password in the
New Password box and the Confirm Password box. Click on the Apply button.

6. Now click the + to expand the Internet Telephony Service Provider name (e.g. Voiceflex) and
then click the + to expand the Internet Telephony station added in step 4 (e.g. 84411230).

There are two ways to add MSN/DID numbers to the ITSP account, if the customer only has a
few DID numbers available from the ITSP or the numbers are not consecutive proceed to step
6a, if the customer has several consecutive DID numbers provided by the ITSP proceed to step
6b.

a. Right-click MSNs and select Add MSN. Enter the full Internet Telephony Phone
Number (e.g. 01582344277) provided by the ITSP. Select the Internal Call
Number that this MSN will be assigned to from the drop-down list or select ‘insert
number directly’ and enter the station number in the box below.

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NOTE #5 If you intend to allow stations to make outgoing calls via this ITSP account
that DO NOT have an MSN number assigned, the Default Entry flag MUST be
enabled for one of the MSN numbers. Otherwise ONLY stations with an MSN number
assigned will be able to make outgoing calls via this ITSP using this account. Stations
that do NOT have an MSN number assigned will present the MSN number set as
Default Entry as the CLI on outgoing calls.

b. Right-click DID ranges and select Add DID range. Split the DID range provided
to the customer by the ITSP into two parts e.g. The DID range 01582344277 to
01582344283 is available. The first nine digits do not change so this should be
entered as the PABX number, the Direct inward dialing band up would then be
77 (the start of the DID range) to 83 (the end of the DID range). Enter the first
station number in the Begin of internal Call Number range (e.g. 1000), this will
be assigned to the first DID number in the range i.e. 0158234427777. Enable the
Assign direct inward dialing band automatically to stations flag, this DID
range will then be assigned to stations consecutively starting at the station
specified (e.g. 1000). Click on the Apply button.

This will create an individual MSN entry for each DID number in the range and
will assign each MSN number to a station, incrementally, starting with the first
station number entered (e.g. 1000). If an individual MSN needs to be assigned to
a different station afterwards, expand the MSNs tree, right-click the required
MSN, select Edit MSN and change the Internal Call Number accordingly.

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7. Go to the Explorers > Voice Gateway screen. Right-click SIP Parameters and select Edit SIP
Parameters. The Maximum possible Provider Calls value defines the number of
simultaneous calls allowed for all SIP providers. The value entered depends on the bandwidth
available via the customers Internet connection and the codec used. Click on the Apply button.

8. To activate this ITSP right-click the Internet Telephony Service Provider name (e.g.
Voiceflex) and select Activate. An Information window will open; select the Provider
designation in the system from the drop-down list to be the same value as entered in step 4
of the HiPath 3000 configuration (e.g. 1). Click on the OK button.

Click on the Save icon.

9. If the configuration is correct the folder for the Internet Telephony Service Provider name
(e.g. Voiceflex) will change to green. Also, the Internet Telephony station added in step 4
(e.g. 84411230) should also be green indicating that this account has registered successfully
with the ITSP.

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4 Direct Payload

Normally, a DSP (B-channel) is required for an incoming or outgoing call between an ITSP and an IP
workpoint.

From HiPath 3000 V7 R2.0.0 direct payload is supported for Internet telephony via an ITSP. This feature
allows the payload to route via the HiPath HG1500/STMI2 without using a DSP (B-channel) in the
HiPath HG1500/STMI2 card. If an IP workpoint is used no DSP (B-channel) is required, obviously if a
TDM telephone is used 1 DSP (B-channel) is still required.

Therefore the number of DSPs (B-channels) is no longer one of the limiting factors in determining the
maximum number of simultaneous calls possible over the ITSP when using IP workpoints. Instead, with
direct payload the limiting factor is solely determined by the codec used by the ITSP and the bandwidth
available on the broadband Internet connection (for ADSL the upstream speed is the limiting factor as it
is lower than the downstream speed).

The table below illustrates DSP usage for different types of calls. The changes with direct payload are
indicated in brackets.

Called Party

Internal Connections External Connections

Calling Party
TDM Station IP Workpoint
ISDN IP Trunking ITSP
(UP0/E / POT / S0) (HFA / H.323 / SIP)

TDM Station
Connections

0 1 0 1 1
(UP0/E / POT / S0)
Internal

IP Workpoint
1 0 1 0 2 (0)
(HFA / H.323 / SIP)

ISDN 0 1 0 1 1
Connections
External

Not
IP Trunking 1 0 1 0
supported

Not 2 (Not
ITSP 1 2 (0) 1
supported supported)

NOTE #6 With the direct payload feature enabled, transit ITSP calls will NOT be possible. If the
customer needs to connect an incoming or outgoing ITSP call with an outgoing ITSP call either manually
by a user or automatically via call management then the direct payload feature MUST be disabled.

NOTE #7 On HiPath 3000 V6.0 and HiPath 3000 V7 systems running MR1 (V7 R1.x.x) software the
direct payload feature MUST be disabled as it is NOT supported.

4.1 Direct Payload Configuration

To enable the direct payload feature configuration changes must be made within the HiPath 3000 and
the HiPath HG1500/STMI2, proceed as follows.

1. Using HiPath 3000 Manager E disable the Always use DSP flag for the ITSP route (e.g. trunk
group 12). This flag can be found in the Lines / networking object under the Routes tab.

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2. Also, the Use RTP Proxy flag MUST be enabled in HiPath HG1500/STMI2 WBM. This flag can
be found by navigating to Explorers > Basic Settings > Gateway.

3. Obviously to disable the direct payload feature, the Always use DSP flag MUST be enabled
for the ITSP route and the Use RTP Proxy flag MUST be disabled.

NOTE #8 The Extended Fast Connect flag is shown enabled in the screenshot above. This
only applies to HiPath 3000 V7 or later. On HiPath 3000 V6.0 this flag MUST be DISABLED on
ALL HiPath HG1500/STMI2 cards.

4. The number of UDP ports available for the RTP stream may need to be increased. In the
Network object click on the IP Ports tab. Check that the RTP_MIN and RTP_MAX values are
as shown in the screenshot below, and change them accordingly.

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NOTE #9 On HiPath 3000 V5.0/V6.0 the RTP_MAX value was 29131 giving a range of 32
ports for UDP. From HiPath 3000 V7 this value was increased to 29259, giving a range of 160.
However, if the KDS was converted from HiPath 3000 V5.0 / V6.0 the range will still be 32.

5 NAT / STUN

5.1 NAT

Most ADSL routers incorporate a type of Network Address Translation (NAT). NAT is used to allow
multiple hosts’ in a private network access to / from the Internet using one public IP address. However,
NAT breaks end-to-end connectivity because the host in the private network (in this case the HiPath
HG1500/STMI2 card) is not aware of the public IP address.

The SIP protocol operates by embedding network layer (ISO layer 3) and transport layer (ISO layer 4)
information at the application layer (ISO layer 7). When communicating with an ITSP via NAT the
outgoing SIP messages sent by the HiPath HG1500/STMI2 card to the ITSP will contain the private IP
address. The ITSP will reply to this IP address, however, because a private IP address is not routable
the SIP messages and RTP packets will not reach the HiPath HG1500/STMI2 card.

5.2 STUN

STUN (Session Traversal Utilities for NAT) allows the HiPath HG1500/STMI2 to discover the public IP
address and the type of NAT service being used on the ADSL router by communicating with a STUN
server in the Internet. The outgoing SIP messages to the ITSP will then contain the public IP address,
thus allowing the replies from the ITSP to be routed back to the NAT in the ADSL router. Although any
public STUN server IP address can be used it is preferable to use the STUN server provided by the
ITSP.

5.2.1 STUN Configuration

1. Navigate to Explorers > Voice Gateway, right-click on the Internet Telephony Service
Provider heading and select Display STUN Configuration. For most installations the STUN
Mode should be set to ‘Automatic’. The type of NAT detected is also displayed.

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2. To change the STUN Mode, right-click on the Internet Telephony Service Provider heading
and select Edit STUN Configuration. The following settings are possible:

 ‘Off’ – This disables STUN. This option should only be used if the ADSL router does
not use NAT.

 ‘Always’ – STUN is always used.

 ‘Automatic’ – The system will attempt to detect when STUN is required (default)

 ‘Use static IP’ – STUN is disabled. This option allows the Public IP Address and
Public SIP Port (default 5060) to be specified.

 ‘Port Preserving Router’ – STUN is disabled. This option can be used if the NAT
internal port is always the same as the external port.

Click on the Apply button.

NOTE #10 Even if STUN is disabled (‘Off’, ‘Use static IP’ or ‘Port Preserving Router’) it is
important that a STUN server is configured for the ITSP (section 3.2 step 4). If the entry is left
blank the HiPath HG1500/STMI2 will use the STUN server for the first ITSP configured.

3. After adding a STUN server (section 3.2 step 4) or enabling STUN (‘Always’ or ‘Automatic’) it is
wise to start the NAT type detection again, right-click on the Internet Telephony Service
Provider heading and select Detect NAT Type.

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6 Troubleshooting

6.1 ITSP will not register

There can be many reasons why the ITSP will not register, follow the checklist below:

1. Check the HiPath 3000 and HiPath HG1500/STMI2 configuration again (section 3), step-by-
step and check that the configuration matches the screenshots provided. Most ITSP related
TAC calls are caused by not following ALL the steps or reading all of the notes.

2. Try and ping the ITSP registrar IP address / host name (as entered in section 3.2 step 4) from
the HiPath HG1500/STMI2 card. Navigate to Explorers > Routing > IP Routing. Expand the
ICMP Request tree, right-click Ping and select Execute Ping. Enter the IP address or host
name into the Target Address box and either click on the Send or Send (in a separate
window) button as required.

If the ping fails, this indicates a basic connectivity problem, check the following:

 Has the correct default router been entered (section 3.2 step 2)? This is usually the
ADSL router. Try pinging the default router from the HiPath HG1500/STMI2 card.
 Try pinging the IP address instead of the host name of the default router from the
HiPath HG1500/STMI2 card (use the nslookup command in a command prompt
window on a PC connected to the Internet to find the IP address e.g. nslookup
sip2.fsys.co.uk), if this works suspect the DNS server entered (section 3.2 step 3).
 If a ping to the default router and DNS server succeeds then the problem may lie with
the ADSL router or the Internet connection.

Make sure that the STUN server IP address has been entered (section 3.2 step 4). Also make
sure that STUN is configured correctly as per section 5.

3. Disable ALG functionality within the ADSL router. Many ADSL routers are supplied with ALG
(Application-Level Gateway) functionality which is enabled by default. While ALG in general
can assist in NAT traversal the implementation is incorrect at the SIP protocol level on many
ADSL routers. The ALG function performs a deep packet-inspection and can actually rewrite
information inside the SIP messages (SIP headers and SDP body) with INCORRECT
information on many ADSL router models. The usual problem is that ALG replaces the public
IP address with the private IP address of the ADSL router in the SIP messages.

6.2 Calls Cutting-Off / Crosstalk / Loss of Payload

Problems have been discovered with the Cisco 800 Series of router when STUN is enabled on the
HiPath HG1500/STMI2. This router is detected as ‘Port-Restricted cone NAT’ by STUN but intermittently
behaves as ‘Symmetric NAT’. If problems are encountered we have found it is best to set the STUN
Mode to ‘Use static IP’ (section 5.2.1). Enter the Public IP Address (this can be found from the ADSL
router configuration or from the ISP). Leave the Public SIP Port in default (5060). This also effectively
disables STUN.
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