VoIP Service Reference

					   IceWarp Unified Communications



   VoIP Service Reference
   Version 10




Printed on 22 October, 2009
                                                                                                                                                                                        i




Contents

VoIP Service                                                                                                                                                                          1

    Introduction ................................................................................................................................................................... 1

    V10 New Features .......................................................................................................................................................... 3

                SIP – REFER ........................................................................................................................................................ 3

                SIP Call Transfer Agent Settings ......................................................................................................................... 3

                NAT traversal - RTCP .......................................................................................................................................... 3

                Video Calls ......................................................................................................................................................... 3

                Multiple Resources- Contact Binding ................................................................................................................. 3

                Incoming Calls To Groups .................................................................................................................................. 3

                Logging - SIP Calls............................................................................................................................................... 3

                Statistics – SIP Calls ............................................................................................................................................ 4

                The Big Picture ................................................................................................................................................... 7

    General ........................................................................................................................................................................... 8

    Gateways........................................................................................................................................................................ 9

    Call Forwarding ............................................................................................................................................................ 13

    Advanced...................................................................................................................................................................... 15

    Using the Dial Via SIP Functionality .............................................................................................................................. 16

    Setting up a SIP Client - X-Lite ...................................................................................................................................... 18

                First run of X-Lite ............................................................................................................................................. 19

                SIP Accounts Dialog.......................................................................................................................................... 21

    Settings for the Grandstream Hardware SIP Phone ..................................................................................................... 25

    Access Mode ................................................................................................................................................................ 27
                                                                                                                                 1



    CHAPTER 1

    VoIP Service

Introduction
    IceWarp Server VoIP Service implements SIP. The SIP (Session Initiation Protocol) is designed to allow devices, both software
    and hardware, to establish a communication session.

    The VoIP Service in IceWarp Server is actually a SIP domain which should be defined within IceWarp Server as a Domain or
    Domain Alias. This Domain must have a valid DNS "A" record.

    The four basic components of a SIP session are:

    SIP User Agents

    These are the end-user devices.

    These can be Software devices, running on PCs, PDAs, Cell phones etc. or they can be SIP-enabled network devices such as
    SIP-phones, or even, via SIP Gateways, ordinary telephony devices.

    A SIP call is initiated by a User Agent Client and responded to by User Agent Server.

    SIP Registrar Servers

    These are databases containing the location of all User Agents within a Domain. There servers retrieve and send IP addresses
    and other information at the request of a SIP Proxy Server

    SIP Proxy Server

    A SIP Proxy Server accepts session requests from a User Agent and queries a SIP registrar for the recipient's address. It then
    forwards the session invitation directly to the User Agent if it is in the same Domain or to another Proxy Server if the User
    Agent is in another Domain.

    SIP Redirect Servers

    These allow Proxy Servers to locate other, external Proxy servers (rather like a DNS for SIP).

    NOTE that in IceWarp Server the Registrar, Proxy and Redirect Servers are integral to the software, no further software is
    required

    The following diagrams and examples should help explain the structure and process of placing a SIP call.
2           VoIP Service Reference IceWarp Unified Communications



    In This Chapter
V10 New Features .................................................................................. 3
General................................................................................................... 8
Gateways ............................................................................................... 9
Call Forwarding ...................................................................................... 13
Advanced ............................................................................................... 15
Using the Dial Via SIP Functionality........................................................ 16
Setting up a SIP Client - X-Lite ................................................................ 18
Settings for the Grandstream Hardware SIP Phone ............................... 25
Access Mode .......................................................................................... 27
                                                                                                                                    3



      CHAPTER 2



V10 New Features

SIP – REFER
      100% RFC compliant, allows SIP Server to be configured with external SIP account by the means of siprefer.dat. Calls can be
      transferred over to remote gateway and back. See the following.



SIP Call Transfer Agent Settings
      Call transfer to any SIP Server (e.g. Asterisk), across SIP servers and gateways, no gateway, account configuration required,
      IceWarp would be used only as call initiator but your real SIP account would be used to make the final call,
      mailbox/~sip/siprefer.dat file, <regex> <sip_destination> on each line containing custom siprefer addresses, RegEx replace
      based.



NAT traversal - RTCP
      RTCP stream support added, off by default. RTP is using an even port number, +1 is used for RTCP. RTCP is used for stream
      quality control and negotiation between clients. Unsupported by common clients, but may be required for video playback.



Video Calls
      Support for video stream has been tested and works with common desktop clients.         The RTP NAT traversal automatically
      assigns 2 more streams for the video RTP.



Multiple Resources- Contact Binding
      Contact list retrieval based on CallID (multiple binding). Allows the same user to connect with several clients at once and
      make calls on each, receive calls on any (rings to all). Very similar to instant messaging resources and concurrent logins.
      Suppose you have a wired phone and you connect with your softphone. At this moment the softphone will take over all calls.
      Once logged out the wired phone will work as usual.



Incoming Calls To Groups
      Incoming calls can be processed by a group or the whole domain if desirable.



Logging - SIP Calls
      Logging of SIP Calls is available in Logs. Administrator can get overview of the traffic over SIP gateway in terms of calls made.
       4       VoIP Service Reference IceWarp Unified Communications




Statistics – SIP Calls
       Statistics - Sessions - SIP Calls added, session termination supported.

       ____________________________________________________________________


       TCP and TLS Calls and Instant Messaging

       SIP and SIMPLE secure connections over SIP TCP and TLS (SIPS) are supported, tested with CounterPath Bria.


       New Rules Functions

       New functions: RESPONSE (allows to send your own response to a SIP request), METHOD (create a RegEx restriction to SIP
       method), STOP (stop processing the SIP packet) and SMS (send text message with parameters e.g. "1" or "maxmsgs=1"),
       integrates smoothly with the SMS service to send SMS over SIP SIMPLE.


       Use Rules Although URI Is Local and Existing Account

       Gateway rules override local URI's (higher priority for local accounts, if call is made to "001" and "001" is local and a a routing
       rule matches "001", the call will be made to the local account anyway.


       00 For + Replacement

       "00" for "+" sign replacement in smsgateways.dat, can be turned off using XML option IGNOREPLUSREWRITE. Many VoIP
       carriers don't support "+" prefix.


       Record-Route

       Response Record-Route rewriting support added. Rewrite and number match in gateway dial rules - all RegEx and RegEx
       rewrite driven, previous compatibility not preserved, requires all rules to be rewritten, e.g. Condition: ^(0)(.*) Rewrite: $2
       (checks if number starts with 0 and rewrites to the remainder), Condition: ^([0-9])(.*) Rewrite: $1$2 (checks if numbers starts
       with any digit and keeps the number - rewrite could be blank to keep the original number).


       Authentication- DIGEST-MD5

       SIP server supports DIGEST MD5 authentication if required.


       e164.org Support

       e164.org is a free Internet service allowing to register phone numbers easily, so that clients can retrieve information about a
       telephone number from DNS, find the real SIP address, email, PSTN phone number etc. In addition to the existing e164.arpa
       lookup, IceWarp SIP has been updated and tested for 100% compliance with e164.org.


       SIP RTP Dump

       Internal option for use with RTP Dump troubleshooting utility. Allows to save streams to RTP Dump, a special format
       supported by WireShark and other tools.
                                                                           Error! No text of specified style in document.      5




RTPDump and RTPDump to MP3

Utility for VoIP SIP troubleshooting. Requires lame_enc.dll, captures PCMU streams, allows to play the stream, convert to
MP3, create call recording by overlaying 2 streams.


Miscellaneous Optimizations

Improved TCP Keep-Alive, Location service (keeps users logged in), Contact: * removal support added. Updated processing of
requests and responses from the Contact: header perspective, local packets bypassed. Init and Timer updated. SIP gateway
startup delayed after service really started. Port 0 handling. Send failure logged. Stream timeout not applied to RTCP. If
External host IP is blank it is automatically detected and set. User verification based on SIPLocation service verifies both email
and IP:port in non-anonymous access. TCP additional bindings - Response finds the proper binding and uses it to send the
packet.




   1.   User A places a call to User B (User B is in a Domain external to User A's Domain). This request is picked up by SIP
        Proxy A. (arrow 1).
   2.   SIP Proxy A determines that User B is outside its Domain so asks a SIP Redirect Server where "User B of Domain B" can
        be found (arrow 2).
   3.   SIP Redirect Server responds with the address for SIP Server B (arrow 2).
   4.   SIP Proxy A sends the call request to SIP Server B (arrow 3).
   5.   SIP Server B requests the location of User B from SIP Registrar B (arrow 4).
   6.   SIP Registrar B responds with User B's location (arrow 4).
   7.   SIP Proxy B contacts User B's device (arrow 5).
   8.   User B accepts the call.
   9.   User B's device tells SIP Proxy B (arrow 5).
   10. SIP Proxy B tells SIP Proxy A (arrow 3).
6        VoIP Service Reference IceWarp Unified Communications


    11. SIP Proxy A tells User A's device (arrow 1).
    12. Channel is established (arrow 6).


If you have multiple Users behind a firewall or router then you will probably need to enable NAT Traversal on the SIP Server
(see SIP - Advanced (see "Advanced" on page 15) tab).

The following diagram shows a call using NAT traversal and Proxy ports.




The basic functionality is the same except that all communication outside of the Domain is done via a proxy port.

One proxy port is opened for each User communicating outside the Domain.
                                                                                Error! No text of specified style in document.   7




      A SIP gateway is a service provided that allows you to connect to non-SIP devices, such as the public telephone network.
      These services usually have to be paid for.




      The initiation of the call is the same up to the point where the SIP Gateway is reached.



The Big Picture
      The SIP server allows you to offer a complete voice communications solution to your Users.
    8            VoIP Service Reference IceWarp Unified Communications




General



        Field                           Description

        Active                          Check this box to enable the SIP server
        Disable anonymous               Check this box if you do not wish to allow anonymous users access to the SIP service.
        access
                                        If you select this option you can use the B button to edit a bypass file, allowing IP ranges, Users
                                        and Domains anonymous access.
        SIP access mode:                Use the Access Mode button to specify which Accounts, Domains etc. will have access to the SIP
                                        Service. This opens the standard Access Mode dialog. See Access Mode for further
                                        information.




        Field                           Description

        Local Network                   Here you need to specify all the local IP addresses that this SIP server should be available for.

        Local interface host            Specify the local IP address of the SIP Server here.
        External interface host         Specify the local IP address of any External interface (probably your router or firewall)

    NOTE - Incorrect Routing information is the biggest cause of problems with SIP communications. Make sure you set this up
    correctly.




        Field                           Description

        Max number of                   Once the maximum number of simultaneous calls is reached any further attempted calls will be
        simultaneous calls              rejected by the Service.
                                        This can be useful if you want to limit the bandwidth that is used by the SIP Server.
                                                                              Error! No text of specified style in document.    9




                                  A typical voice call is around 8kB/s.

    Log user calls to user        Check this box to have SIP calls logged to the mailbox of the user who made the call.
    mailbox
                                  NOTE - This option must be checked for the REDIAL to work.

    Log all calls to file         Check this box and specify a fully qualified path to the file where a log of ALL calls should be
                                  stored.
                                  Note that yyyy, mm and dd can be used in the directory name,
                                  For example:
                                  <InstallDirectory>\SIPyyyymmdd\sip.log




Gateways
   Here you should specify any Gateways you wish to route calls to.

   Gateways are usually an interface to a non-SIP communications system, such as Public Telephony, and you would normally
   have to pay subscription or usage charges to the Gateway provider.

   Selecting the Gateways tab presents you with a list of defined Gateways.




   Use the Delete button to delete a selected Gateway.

   Use the Save and Load buttons to respectively save and load a list of Gateways. A standard file browser dialog will be
   presented.
10        VoIP Service Reference IceWarp Unified Communications



Pressing the Add or Edit button will open the SIP Gateway dialog:




 Field                           Description

 Active                          Check this box to make this Gateway active.
 Title                           A descriptive name for the Gateway.
 Server                          Specify the IP address or hostname of the Gateway.
 Proxy                           Specify the IP address or hostname of any proxy server IceWarp Server should use to get to this
                                 Gateway.
 Username                        Specify the username supplied by your Gateway provider.
 Password                        Specify the password for the above username.
 Expire                          Specify here how often, in seconds, the SIP Gateway should re-register with the Gateway.
                                 Basically, this tells the Gateway that the server is still here and available.
 Max. number of                  Specify here the maximum number of simultaneous calls allowed via this Gateway.
 simultaneous calls
                                 Can be useful in limiting bandwidth usage.
 Rewrite From header             Normally this field should be left un-checked.
                                 This is here in case your gateway provider requires it.
 Rewrite To header               Normally this field should be left un-checked.
                                 This is here in case your Gateway provider requires it.
                                                                           Error! No text of specified style in document.   11




The Outgoing Calls tab allows you to select particular calls to use this Gateway, using the number prefix.

A number prefix must be assigned to a Gateway, even if there is only one Gateway, or the Gateway will be inaccessible.

If you have multiple gateways then this is effectively a way to select which gateway the call will be routed through. You may
have different SIP to phone providers for different countries and want to route calls appropriately. You can select you own
prefix for each gateway and publish the list within your organization.


 Field                         Description

 Number prefix                 Specify the number prefix that will cause a call to be routed through this Gateway.
                               The above example shows that any recipient address that starts with 555 will be processed via
                               this Gateway.
                               Multiple prefixes can be entered, separated by semicolons.

                               NOTE - do not enter any spaces in this field as this will cause a failure.

 Replace number                This field allows you to modify the prefix if necessary.
                               You can enter numbers here and there are three special stings that are substituted by part of the
                               original dialed number:
                               %& - is substituted with the original number, minus the prefix.
                               %^ - is substituted with just the prefix
                               %* - is substituted with the whole number
12        VoIP Service Reference IceWarp Unified Communications


                                  These stings can be selected via the button to the right of the text field.
                                  Example (see previous screenshot)
                                  This gateway is a SIP to Phone gateway offering cheap calls to the UK, but the gateway requires
                                  that the full UK number is specified for the call (without the country code but with the leading
                                  zero).
                                  Specify Number prefix as 0044;44;+44
                                  Specify Replace number as 0%&
                                  and if a user dials +4479795551234, the call would be routed to 079795551234 via this gateway.
 Restrict calls to users          Check this box to restrict the Users who can use this Gateway to place an outgoing call.
 below
                                  This option is recommended as otherwise you could be leaving your Gateway open to abuse by
                                  anyone that knows of its existence.
                                  Use the Add button to open the Select Item dialog, allowing you to add Accounts and/or
                                  Domains to the list.




The Incoming Calls tab allows you to specify where an incoming call to you number on this Gateway is routed.

Use the Add button to specify which User or Users the call should be routed to.

     If no Users are specified IceWarp Server will do nothing with the call request, not even reject it.
     If one User is specified then IceWarp Server will attempt to route the call to that User.
     If multiple Users are specified IceWarp Server will attempt to contact all of those Users simultaneously, and will wait until
          either:
        A User accepts the call, in which case it is routed to that User
                                                                                 Error! No text of specified style in document.   13




           All Users reject the call or the request times out, in which case IceWarp Server will reject the incoming call

    Use the Delete button to delete a selected Gateway




Call Forwarding
    The Call Forwarding tab allows you to specify rules on how to handle SIP requests, Accept, Reject or Drop based on where
    the request is coming from, going to, etc.

    You are presented with a list of defined rules:




    Away number

    The Away number is a number which any user can dial to have his phone "switched off" or forwarded to another number, by
    the server.

    The user initiates this on demand from his own SIP device, and IceWarp Server instantly writes a Call Forwarding rule for the
    number.

    Specify here the number a User should dial.

    Reset number

    Specify here the number a User should dial to "switch on" his phone and cancel any Call Forwarding.

    In the above example:

           if a User dials *151* then the server will reject any calls to that User.
           if a User dials *151*3105551234 then the server will forward calls for that user to 3105551234
           if a User dials *151*john@icewarpdemo.com then the server will forward any calls to john@icewarpdemo.com

    NOTE that the above Call Forwarding is dynamic, initiated and cancelled by individual Users via their SIP device. The Call
    Forwarding described later in this section is controlled by the system administrator, and is not dynamic.

    Redial number
14       VoIP Service Reference IceWarp Unified Communications



Specify here a number that a User can dial to redial the last number for his address.

NOTE - that the number dialed will be the number associated with the last event associated with this User, whether it is an
outgoing or incoming call.

Use the Delete button to delete a selected rule.

Pressing the Add or Edit button opens up the Call Forwarding dialog:




 Field                          Description

 Number                         Specify here the original intended recipient of the call.
                                Multiple recipients can be specified, separated by semicolons.
                                Masks can be used, using %, instead of *, to specify any string. (This is because * is valid in
                                telephony terms)
                                Examples:
                                 +11% - means any string starting with +11.
                                %@icewarpdemo.com - means any string ending with @icewarpdemo.com.
                                %domain% - means any string containing domain.
 Forward To                     Specify here a number or address for the new recipient.
                                Multiple addresses/numbers can be specified, separated by semicolons.
 Description                    Short descriptive text for identification purposes.
                                                                            Error! No text of specified style in document.    15




Advanced
   The Advanced tab allows you to specify how the SIP server will perform NAT translation, and whether the server will use a
   parent SIP proxy.




    Field                        Description

    Use Telephone/E164           Check this box to enable your users to place calls to standard telephone numbers.
    Number Mapping
                                 The ENUM system allows a standard telephone number to be dialed from a SIP client.
    (ENUM)
                                 The SIP server will check for a NAPTR DNS record based on the number dialed (the 164.arpa
                                 server is tried out, if record not found, then the e164.org one.)

                                 NOTE that the option mentioned above has higher priority than SIP gateways (if used.)

    Use extended DNS             Select this option to have IceWarp Server check for SRV and NAPTR DNS records to determine
    lookup (NAPTR and            the hostname for a SIP Server.
    SRV)
                                 IceWarp Server first checks for a NAPTR DNS record and , if none is found, it will check for an SRV
                                 DNS record.




    Field                        Description

    Enable SDP NAT               Check this box to enable the SDP (Session Description Protocol) NAT (Network Address
    traversal proxy              Translation) Proxy.
                                 NAT is used to correctly route incoming data to the correct local network recipient.
    RTP NAT Traversal            RTP (Real-time Transport Protocol) is the protocol used by the SIP server for streaming data.
    server mode
                                 The SIP server can create Proxy Ports for each SIP call. This is useful if you have no control over
                                 the ports being used by your User's SIP clients.
                                 Choose from the following options:
                                 Disabled
    16       VoIP Service Reference IceWarp Unified Communications


                                      Select this option and no Port Proxies will be created
                                      Sessions from external hosts
                                      Select this option and Port Proxies will be created for communications with external hosts.
                                      All sessions
                                      Select this option and Port Proxies will be created for all SIP server communications. This is
                                      especially useful if you have many Windows XP Users.
     Local RTP port range             You need to specify the Ports to be used as Proxies by the SIP server.
     from
                                      You should specify the start of the port range to be used
     Local RTP port range to          Specify the last port of the range to be used.

                                      NOTE that the port range specified here must be open in your router/firewall setup.




     Field                            Description

     Use parent SIP proxy             You can have multiple SIP servers with only one server having access to the outside world.
                                      In this case you would specify this external server as the "parent" server.
                                      Check this option if you wish to use a parent server, and specify the hostname of the parent in
                                      the text box
     Parent proxy                     Specify the IP address or hostname of the parent SIP server.
     Contact: registration            Specify a value here to tell clients that they should re-register with the server at the interval
     expiration (Sec)                 specified.
                                      This can be very useful to keep the Client/Server connection alive.




Using the Dial Via SIP Functionality
    Both the Outlook Connector and IceWarp WebClient have the ability to dial out via SIP clients.

    In the Outlook Connector

            Locate and select the contact you wish to call (if the person you wish to call is not in your contacts skip this step)
            Select "Dial via Merak Server" from the Outlook Connector dropdown menu.
            Check the correct contact is displayed and click Dial
            Your SIP client will start to ring, answer it.
            After a couple of seconds the other person's SIP client will be contacted and your conversation can start.
                                                                           Error! No text of specified style in document.   17




In IceWarp WebClient

      Click "Dial" on the menu bar.
      Select the Contact you wish to call (or type in the email address) and click Dial.
      Your SIP client will start to ring, answer it.
      After a couple of seconds the other person's SIP client will be contacted and your conversation can start.

Note that the call is in no way routed by Outlook Connector or IceWarp WebClient, they are just used to initiate the call. The
SIP server dials your registered client and once connected route the call to the destination you specified. This method will
work with any SIP client.
                                                                                                                  18



     CHAPTER 3



Setting up a SIP Client - X-Lite
     There are numerous SIP clients available, both software and hardware.

     This section describes how to set up X-Lite to access your SIP server.

     X-Lite is available from http://www.counterpath.com/


       In This Chapter
     First run of X-Lite.................................................................................... 19
     SIP Accounts Dialog ................................................................................ 21
                                                                              Error! No text of specified style in document.   19




First run of X-Lite
      When you first run X-Lite it will discover that you have no SIP account defined and will show the message "No SIP accounts
      are enabled""




      And the "SIP accounts" dialog will we be displayed automatically.
20      VoIP Service Reference IceWarp Unified Communications



If you wish to add your IceWarp Server account to X-Lite you should open the "SIP accounts" manually:
                                                                              Error! No text of specified style in document.   21




SIP Accounts Dialog
     The SIP Accounts dialog shows you a list of all the SIP accounts you have defined.
22       VoIP Service Reference IceWarp Unified Communications



Press the Add button to define your IceWarp Server account. The properties dialog will be displayed:




 Field                          Description

 Display name                   Enter the display name you would like people to see when you are in a call with them.
 User name                      The User name supplied by your SIP service provider.

                                NOTE that it is necessary to log in using this user name only (not the user's e-mail address). In case of
                                IceWarp Server use the alias of the appropriate account.

                                NOTE that this applies even if the "Users login with their email addresses" option (Domains &
                                Accounts / Policies / Login Policy / Login Settings) of the IceWarp Server is selected.

 Password                       The password supplied by your SIP service provider.
 Authorization user             Same as your User name.
 name
 Domain                         Enter the domain name of your SIP service supplied by your SIP service provider.
 Domain Proxy                   Leave as is.
                                                  Error! No text of specified style in document.   23




 Dialing plan                      Leave as is.

Press OK to return to the SIP Accounts dialog:




Press Close to return to X-Lite.
24      VoIP Service Reference IceWarp Unified Communications



X-lite will now attempt to connect to your SIP service provider and will show the following if successful:
                                                                         Error! No text of specified style in document.   25




Settings for the Grandstream Hardware
SIP Phone
    The following screenshot shows the settings for the Grandstream hardware SIP phone:
26        VoIP Service Reference IceWarp Unified Communications




The settings you need to change for IceWarp Server are:

     Proxy - The SIP server domain name
     Display name - the name you wish others to see when you are in a call
     Password - the password for your SIP server
     Auth ID - Your SIP server username
                                                                            Error! No text of specified style in document.    27




Access Mode




   The Access Mode lets you specify which accounts are allowed to access the service.


    Mode                         Description

    All accounts                 The service is accessible by all accounts in all domains on this server.
    Accounts from list           Only accounts/domains listed in the text box can access the server
                                 Enter the accounts that are allowed to access the service in the List text box, separated by
                                 semi-colons.
                                 Use the '...' button to open the Select Item dialog to select accounts.
    Use domain options           Only accounts in domains that have the service selected in Domain Options can access the service.

    Use account options          Only accounts that have the service selected in User Options can access the service.

    Advanced mode                Access will be granted to all accounts which have access:
    (Logical NOT XOR)                   Disabled via both Domain Options and User Options.
                                         or
                                        Enabled via both Domain Options and User Options.
                                 Example:
                                 Backup domains do not usually have users but they can have. By default, all backup domain users
                                 (both local and locally non-existing ones) have services (e.g. anti-spam) enabled. You can want to
                                 use this service just for local users. It is possible to use the "Accounts from the list" mode but it is
                                 not too handy. Better solution is to use the "Advanced mode", deselect the service on the domain
                                 level and on the user level deselect the service for all local users. (It means that they will have the
                                 service enabled. Alternatively, you can create a user template with this service deselected and use it
                                 as a default one.)
    List Accounts...             Clicking this button reveals the list of accounts or domains currently enabled for the service:

                                        In the 'Use domain options' mode – the current list of domains.
                                        In the 'Use account options' and 'Advanced mode' modes – the current list of users.

                                 Note- that in the 'All accounts' and 'Accounts from list' modes – this button is disabled.
28   VoIP Service Reference IceWarp Unified Communications

				
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