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StarTrinity VoIP Softswitch Tutorial

Database architecture diagram

The database architecture diagram is the best way to show you concepts and features of our softswitch. There diagram contains alternative names of entities, because there are multiple names for basic concepts in the VoIP industry
Invoices(payments, money movements)
  • originator ID
  • payment date
  • amount, currency
  • status
  • service, payment details
Invoice Templates
  • ID
  • template
  • detail level
Originators / Customers(Accounts / Calling cards / Clients / Companies)
  • ID
  • parent customer ID
  • routing group ID
  • service plan ID
  • max depth of child customers
  • commission size
  • enabled
  • time zone
  • currency
  • balance, max. balance
  • max. credit
  • enable CDR reports
  • enable tariff change notifications
  • billing email
  • technical email
  • company info: name, address, VAT ID, etc.
  • web auth. info, last web auth. date
  • webAPI auth. info, last webAPI auth. date
  • webRTC API auth. info
  • VoIP auth. info
    (IP address(es), user+password)
  • PIN code
  • tech. prefix, CLD auth. mask (DID)
  • last call date
  • codec configuration
  • disconnection reason mapping
  • max. channels count
  • max. call duration
  • prefix to add/remove,
    CLI/CLD translation rule
  • hide own CLI
  • enable call recording
  • block routing loops
  • media proxy mode: auto, always, never
  • packet capture mode: off, SIP, SIP+RTP
  • status / error message
Service plans
  • ID
  • tariff ID
  • billing cycle: week, month, none
  • invoice template ID
  • billing day: [of week/month], on assignment
  • low balance alarm threshold
  • low balance alarm mode: email, sound during call
  • additional service charges
Tariffs / Destination sets(Pricelists)
  • ID
  • enabled
  • applied to: originator, terminator
  • currency
  • grace period / free seconds
  • connection fee
  • min billed duration
  • max loss % (for originators)
Tariff rates / Routes
  • tariff ID
  • enabled
  • prefix
  • min, max digits
  • country, region (automatically detected)
  • rate1, period1, rateN, periodN
  • activation, expiration date
  • priority in routing group (for originators)
Currency exchange rates
  • currency
  • usd per unit
  • when updated
  • source of data
    • manual
    • yahoo.com
    • XE.com
    • custom server
Routing groups
  • ID
  • routing policies (ordered list):
    • prefix length
    • least coast
    • weighted distribution (load balancing)
    • routing group item order
    • route priority
    • combination of
      ASR, ACD, PDD
Routing group items
  • ID
  • routing group ID
  • destination set ID
  • terminator ID
  • weight
  • order in routing group
  • activation, expiration date
Suppliers(Vendors / Companies)
  • ID
  • company information
  • web auth. info
  • balance
  • base currency
  • time zone
Terminators / Gateways
  • ID
  • supplier ID
  • enabled
  • host / IP address, SIP port
  • IP address is dynamic (behind NAT)
  • VoIP auth. parameters
  • max. channels count
  • max. call duration
  • prefix to add/remove,
    CLI/CLD translation rule
  • CLD, CLI regular expression
  • custom CLI
  • 100, 18x, 200 response timeouts
  • codec configuration
  • media proxy mode: auto, always, never
  • SIP headers proxy mode
  • packet capture mode:
    none, SIP, SIP+RTP
  • hunt stop SIP codes
  • quality monitoring
    • ACD, ASR, PDD thresholds (optional)
    • statistics window
    • bad quality action:
      make last in route, disable
    • notify on status change
    • retry interval
    • retry batch
  • status / error message
Calls / CDRs
  • SIP Call-ID
  • originator ID
  • terminator ID
  • CLI (caller ID)
  • CLD (called number)
  • created, answered, destroyed time
  • VoIP quality fields
  • billed duration
  • status / error message
  • many other fields
Tickets
  • ID
  • created date
  • status
  • SIP Call-ID
  • originator ID (who reported)
Custom tables from your database according to custom CallXML script
Tenants / Softswitch owners / Hosted customers
  • ID / email
  • payment processor API settings: PayPal, skrill
  • web administrator password
  • max. channels count
  • custom CallXML script
  • white label logo settings
System settings
  • name
  • value
System log messages
  • severity level
  • date
  • message

Runtime operations and commands, dynamically calculated fields

Originators / Customers(Accounts / Calling cards/ Clients / Companies)
  • generate invoice
  • complete invoice (add funds)
  • make test call
  • check without call: check routing configuration, get rates, limits, CLI/CLD translations
  • traffic statistics
  • number of child accounts / customers / calling cards
  • export / import accounts
  • analyse impact of network delays on billing
Service plans
  • traffic statistics
Tariffs(Destination sets / Pricelists)
  • export / import rates
  • number of routes
  • traffic statistics
Routing groups
  • automatically generate tariff for originator
Routing group items
  • traffic statistics
Suppliers(Vendors / Companies)
  • traffic statistics
Terminators / Gateways
  • traffic statistics
  • analyse impact of network delays on billing
Calls / CDRs
  • terminate current call
  • export / import CDRs
  • query CDRs by syntax expression. e.g. bad / failed calls: SIP error code, dead air audio, one way audio, high jitter, high packet loss, low MOS, small billed duration
Traffic statistics applied to any subset of calls from CDR
  • grouping by originator, terminator, prefix / route, etc
  • ASR, ACD, PDD
  • number of current calls: ringing, connected, total
  • real capacity chart for terminator based on concurrent calls count and error responses
  • sales / cost / profit
  • billed duration
  • history charts
  • minimum, maximum, average
  • moving averages: based on time, based on N recent calls

Database entities and system concepts

Originators

Originators are ways to receive incoming VoIP traffic to this server. The 'Originator' concept is same as 'Client', 'User', 'Calling card'.
FieldDescriptionExample
IdUnique identifier. Can't be modified after creation
EmailCustomer's email
Customer nameFirst name and last name or name of organization
AddressCustomer's address
EnabledIs used to temporarily disable incoming traffic from originator
Parent originator IDIs used to store default settings for child originators: routing group and service plan
Max channels countLimits number of concurrent calls (channels) from client through this originator account
Max calls per secondLimits number of calls per second (CPS) from client
CallXML variablesList of custom CallXML variables for custom CallXML scriptsvar1=x;var2=y
Authentification settings
Authorize by user and passwordSelects authentication mode: user/password (digest authentication) or IP authentication
Tech. prefixIs used for IP authentication mode. Technical prefix - a required prefix in CLD which is used to authenticate the call. The tech. prefix is removed after authentication, it should not exist in tariff rates list
Auth. IP address(es)Is used for IP authentication mode (client's signaling IP). Semicolon-delimited list of allowed originator's IP addresses. Client's media IP address is not restricted
Auth. user nameUser name for digest authenticaton mode. If set, overrides CLI. Is also used as CLD for free calls between originators (configured in service plan settings)
Auth. passwordPassword for digest authenticaton mode
Registration statusIndicates registration status of SIP phone: online/offline, its IP address
RTP audio settings
Forced codecIf set, specifies only one declared codec in SDP answer. You can see details about codec negotiation in CDR - call details - SIP trace
Desired codecIf set, specifies preferred codec (top codec in list of supported codecs). The setting is used when you want to select specific codec, if possible, and if you still want to support all other codecs
Don't proxy RTP mediaRTP relay mode, is used for open RTP routes. If set, RTP media goes directly from originator to terminator, without passing the softswitch. You might need it to improve audio quality - reduce roud-trip time (RTT, audio latency) and to reduce CPU lo0ad and bandwidth on the softswitch. Drawback of this setting is that originator and terminator can see their RTP IP address and connect to each other without you (as wholesaler). When terminator is your own gateway or server, you can use this setting safely. When you need to convert codecs on the softswitch (transcode), process RTP, measure audio signal or record sound to WAV files, you can not use this setting
Missed call routeIncreases selling capacity for missed call (CC) VoIP routes. Rejects call leg B after first ring or immediately after answer. Fake ringback tone can be played to call leg A to extend alerting time
Record RX streamsTurns on recording of received audio streams to WAV files. The recording could be downloaded from CDR - call details
Record TX streamsTurns on recording of transmitted audio streams to WAV files. The recording could be downloaded from CDR - call details
Record mixed streamsTurns on recording of mixed (received + transmitted) audio streams to WAV files. The recording could be downloaded from CDR - call details
LanguageIdentifies folder for IVR files
DID forwarding settings in DID forwarding mode the call is accepted from DID provider, actual destination number and PIN code are entered via IVR (optionally)
Auth. CLDDID number, is used to authenticate and accept calls from DID. Can be set for this originator or for parent originator
Auth. PIN codePIN code, is used to authenticate call from DID
Auth. CLI'sCallerID's for authentication. One item or semicolon-delimited list. Is used in combination with 'Auth. CLD' for PINless dialing18001234;79123456
Unconditional DID forwarding setting in unconditional DID forwarding mode the call is accepted from DID provider and immediately forwarded to predefined number (usually long-distance number)
Incoming CLDDID number, is used to authenticate and accept calls from DID
Forward to this numberNumber that will go to termination side
Web authentification settings settings for self-care web portal for the client. At the portal the client can view connection settings, CDR (call history), rates, CDR statistics. He can also pay to your PayPal account using PayPal or credit card
Link for originatorHTTP URL to be used by the client
Web login
Web password
Routing
Routing groupSpecifies how the call is routed to terminators. Default value is taken from parent originator. See routing groups
1. Prefixes to removeSemicolon-delimited list of prefixes that have to be removed from original CLD+;00
2. Min digits (inclusive)Minimal number of digits in CLD at this step
3. Prefix to addPrefix to be added to CLD at this step
Status code mappingSpecifies translation of SIP status codes from termination side to origination side, in this format: [oldCode1]-[newCode1];[oldCode2]-[newCode2]... The setting is needed when your clients complain about specific SIP status codes received from the softswitch and if you want to send different response codes instead603-503;404-503
Custom CLISpecifies a custom CLI, overwrites original CLI
2xx (connection) timeoutCall alerting timeout, is used to release hang calls in alerting state
Billing
Service planSpecifies VoIP service plan for the client. Service plan includes tariff and IVR settings. Default value is taken from parent originator. See service plans
Balance (USD)Current balance on client's account. Is updated in real time with live traffic. Positive balance means prepaid billing. Negative balance means postpaid billing, it becomes debt from client to you. Current calls are instantly disconnected when balance goes below 'minus Max credit'
Max credit (USD)Maximal allowed debt for the client. Max credit = 0 means prepaid billing. Max credit = 50 means that balance can go down below zero to -50 (postpaid billing and maximum debt = 50)

Invoices

Invoices of originator represent his history of payments, both completed and incomplete
FieldDescription
CreatedDate and time when the invoice was created
PaidDate and time when the invoice was paid
Amount (USD)
Service descriptionShort description of the VoIP service for the client. May include number of minutes, destinations. CDR file with details could be downloaded at 'CDR statistics' web page
Payment descriptionInformation about payment (e.g. transaction ID) for your own future reference
Balance (USD)Balance of originator (client's account) after the payment

Service plans

Service plans specify VoIP service contracts and IVR settings for originators
FieldDescription
IdUnique identifier. Can't be modified after creation
Tariff IdSpecifies which tariff is used for the service plan
Enable free calls between originatorsEnables calls between VoIP origination clients (SIP softphone users), the calls are not charged. Destination user is found by mathing called number (CLD) to auth. user name
Enable balance IVRWhen set, the softswitch plays originator's current balance to caller before actual call
Enable out of balance IVRWhen set, the softswitch plays a warning to originator when he is dialing and out of balance. Otherwise the incoming call is rejected with erroneous SIP status code
Enable remaining time IVRWhen set, the softswitch plays remaining time before actual call - how much time the caller is able to talk using his current balance
Enable busy IVRWhen set, the softswitch plays audio message when destination number is busy, so it is clear to client why call has failed

Tariffs / Destination sets

Tariffs are lists of country codes (prefixes) and prices that are used by you to bill clients (for incoming traffic) (rates for your customers).
Destination sets are lists of country codes (prefixes) and prices that are used by your suppliers to bill you (for outgoing traffic) (rates of your suppliers/vendors).
FieldDescription
IdUnique identifier. Can't be modified after creation

Tariff rates / Routes

The tariff rate or route is selected based on called number (CLD) without tech. prefix. The softswitch finds rate with longest matching prefix, as long as possible. Example: if you have rates 234 = 0.035, 2347 = 0.04, 23472 = 0.05, then number 23411111111 will be billed as 0.035/minute, 23471111111 = 0.04/minute, 23472111111 = 0.05/minute.
FieldDescriptionExample
PrefixPrefix in called number (CLD), country code with operator code. The prefix can't be edited after creation of the record
DescriptionDescription of the prefix, e.g. country, operator, cityCameroon MTN
EnableIs used to disable traffic to specific prefixes (e.g. premium numbers or blacklisted numbers)
PriorityIs used to elevate priority of the route in the destination set against other destination sets, to select specific terminator for specific prefix. See routing group - routing policy "rate priority + least cost"
Max channelsIs used to limit number of concurrent calls going via the softswitch using this particular rate in destination set
RateRate to bill the calls, in USD/minute. This rate is effective between 'effective from' and 'effective to' dates. Actual current rate depends on current date and time
Period1Minimal billed duration (first/initial billed period), in seconds
PeriodNDuration of consecutive (N-th) billed period, in seconds
Example 1: Period1 = 60 seconds, PeriodN = 60 seconds: call with duration = 90.5 seconds is billed as 120 seconds
Example 2: Period1 = 60 seconds, PeriodN = 60 seconds: call with duration = 2.9 minutes is billed as 3 minutes
Example 3: Period1 = 1 seconds, PeriodN = 1 seconds: call with duration = 90.5 seconds is billed as 91 seconds
Example 4: Period1 = 60 seconds, PeriodN = 1 seconds: caller is initially billed for 60 seconds, after that it goes as per-second billing
Effective fromInclusive starting date to use the 'Rate' field. Before the starting date 'Previous Rate' is used for billing
Effective toExclusive ending date to use the 'Rate' field. Since the ending date 'Next Rate' is used for billing
Previous rateRate to be used before the 'Effective from' date
Next rateRate to be used on and after the 'Effective to' date
Min/max digitsOptional filter for CLD number. Inclusive minimal and maximal number of digits in called number

Suppliers

Suppliers are companies grouping terminators. The 'Supplier' concept is same as 'Termination partner company', 'Provider', 'Vendor'
FieldDescription
IdUnique identifier. Can't be modified after creation
Company name
Debt (USD)Amount of money owed to supplier. Is updated in live mode with live traffic
Web UI user nameAuthentication info for supplier's self-care web portal where he can check his CDR statistics and call history
Web UI passwordAuthentication info for supplier's self-care web portal where he can check his CDR statistics and call history

Terminators

Terminators are ways to send outgoing VoIP traffic from this server. The 'Terminator' concept is same as 'Termination point', 'GSM gateway', 'Device', 'Group of SIMs and channels/lines at GSM gateway'. Terminators are grouped in suppliers (companies who own terminators).
FieldDescriptionExample
IdUnique identifier. Can't be modified after creation
SupplierOptional link to supplier who owns this terminator. Supplier groups multiple terminator that belong to one company
EnabledIs used to temporarily disable outgoing traffic to originator
Enabled untilIs used to temporarily disable outgoing traffic to originator until specific date/time
Max channels countCall capacity of the terminator, maximal number of concurrent channels
SIP connection settings
IP address is dynamicIs set when termination gateway is behind firewall or does not have a static IP address. In this mode the gateway sends REGISTER to the softswitch using user name and password (see below)
Auth. userUser name for authentication for REGISTER requests from terminator/gateway. If set, overrides CLI
Auth. passwordPassword for authentication for REGISTER requests from terminator/gateway
IP address / host nameIP address or host name of destination VoIP provider or gateway's static IP
SIP portSIP port number at destination host. Default is 5060
RTP audio settings
Forced codecIf set, specifies only one declared codec in SDP offer. You can see details about codec negotiation in CDR - call details - SIP trace
Desired codecIf set, specifies first codec in list of all supported codecs in SDP offer (top codec in list of supported codecs). The setting is used when you want to select specific codec, if possible, and if you still want to support all other codecs
Don't proxy RTP mediaRTP relay mode, is used for open RTP routes. If set, RTP media goes directly from originator to terminator, without passing the softswitch. You might need it to improve audio quality - reduce roud-trip time (RTT, audio latency) and to reduce CPU lo0ad and bandwidth on the softswitch. Drawback of this setting is that originator and terminator can see their RTP IP address and connect to each other without you (as wholesaler). When terminator is your own gateway or server, you can use this setting safely. When you need to convert codecs on the softswitch (transcode), process RTP, measure audio signal or record sound to WAV files, you can not use this setting
Record RX streamsTurns on recording of received audio streams to WAV files. The recording could be downloaded from CDR - call details
Record TX streamsTurns on recording of transmitted audio streams to WAV files. The recording could be downloaded from CDR - call details
Record mixed streamsTurns on recording of mixed (received + transmitted) audio streams to WAV files. The recording could be downloaded from CDR - call details
Custom ringback tonesSemicolon-separated list of file names. Audio files are uploaded in audio files manager
Timeout settings
18x (alerting) timeoutOptional timeout to receive 'alerting' response from terminator, in seconds. Is used to release hang calls and try next terminator in routing group
2xx (connection) timeoutOptional timeout to receive 'connected' response from terminator, in seconds. Is used to release hang calls and try next terminator in routing group
Max connected timeOptional max. duration of calls, in minutes. Is used to automatically release hang connected calls
Routing, number validation / filtering, number manipulation
1a. Allowed prefix(es)Is used to validate dialed number, before adding tech. prefix00592;00593;592;593
1b. Regular expression(s)Is used to validate dialed number, before adding tech. prefix.^592[0-9]{7}$;^593[0-9]{7}$
for numbers like 237672991443 (prefix 237 + 9 digits) it would be ^237[0-9]{9}$
2. Prefixes to remove+262;262;00262
3. Prefix to add (tech. prefix)Technical prefix. Is not related to destination set (rate sheet) because rate is calculated before adding the technical prefix
Number validation / filtering for CLD (called ID) and CLI (ANI, caller ID)
4.1 CLI regular expression(s)Is used to validate caller ID^1[0-9]{10}$;^563[0-9]{8}$
^[0-9]+$ (only digits)
4.2 pass CLI only with these numbers of digits10;11;12
4.3 do not pass CLI from this listStatic CLI (caller ID) prefixes or numbers blacklist. The lists of numbers are uploaded on a special webpage
4.4 pass CLI only from this listStatic CLI (caller ID) prefixes or numbers whitelist
4.5 pass CLD only with these numbers of digits10;11;12
4.6 do not pass CLD from this listStatic CLD (called ID) prefixes or numbers blacklist
4.7 pass CLD only from this listStatic CLD (called ID) prefixes or numbers whitelist
CLD dynamic blacklists (RAM-based)
4.8.1 (stage #1) don't pass CLD if there are more than XX attempt(s) in previous YY minute(s)
4.8.2 (stage #2) don't pass CLD if there are more than XX attempt(s) in previous YY minute(s)
5. Change CLI to thisA number to overwrite caller ID
Number of times to retry on 503Number of times to retry call to the terminator if received SIP status code = 503 (Service Unavailable). Is used to increase ASR when terminator (supplier) oversells his capacity
Enable 3xx redirectionEnables redirection when termination side (redirect server) sends response with status 3xx and new destination IP address
Inner CallXML element nameSpecifies name of inner CallXML element name (inside 'transfer' element) when using custom CallXML script

Routing groups

Routing groups are used to link terminators and destination sets and to group them for routing. Example explaining how the routing group works:
You have 3 routes with 3 different rates for the same termination and you want the calls to go to route1 rate1 first; if it fails you want it to try route2 rate2; if it fails you want it to go route3 rate3.
The configuration of routing group is following: the 3 routes mean 3 different terminators going to same IP address but having different tech. prefixes. Each terminator would have its own destination set (pricelist, prefixes with prices). In the routing group you link the 3 terminators and 3 destination sets, and routing would be done according to policy of the routing group ("routing group item order" = try one terminator after another as they are listed in group).
FieldDescription
IdUnique identifier. Can't be modified after creation
Allow negative marginWhen checked, the softswitch blocks calls if termination price is higher than origination price (to avoid profit loss in the softswitch, negative margin)
Routing policySpecifies method of routing calls to list of terminators, order of terminators in routing. One of following values:
  • Routing group item order - terminators are ordered according to order of items in routing group
  • Least cost (LCR) - terminators are ordered according to rate per minute in destination sets, starting from lowest rate
  • Rate priority + least cost - terminators are ordered first by priority of rate in destination sets, then by rate. Is used when you want to change LCR routing, select a specific terminator to be used instead of low-quality but cheapest route
  • Weight - calls are distributed between terminators according to weight of items in routing group. Is used for load balancing, when you need to distribute load between multiple terminators. Also the weight-based routing could be used for CLI/NCLI blending (you must know that the blending is a fraud)
  • Least cost + weight - same as "least cost"; in addition to that calls are distributed between terminators according to their weight, if they have same price

Routing group items

Routing group items are used to link terminators and destination sets
FieldDescription
EnabledIs used to temporarily turn off the routing group item without removing it
OrderOrder of terminator in routing for routing policy "routing group item order"
TerminatorTerminator which is linked to the routing group item
Destination setDestination set which is linked to the routing group item
Max channelsOptional capacity limit for VoIP traffic going via this particular item in this routing group
WeightWeight of the item in load balancing mode (routing group policy "Weight")

Audio files

Audio files are used to play IVR messages and custom ringback tones. The files could be downloaded or uploaded via web interface. WAV and MP3 formats are supported

Number lists

The lists of numbers could be used as white lists or black lists to filter VoIP traffic. A list can contain few million numbers

Web interface

Web pages to configure database entities

The softswitch web interface contains web pages to configure all entities listed above - add new record, view existing records, edit record, delete record. Here is screenshot for originators:

Status

The status screen displays live information about current calls, originators, terminators

CDR

The CDR screen displays call history: list of calls with CDR fields and links to call details page. Both A and B (incoming and outgoing direction) call legs are displayed in the CDR. The calls are sorted by destroyed time, descending (newest to oldest). There are various filters for the displayed calls. CDR fields are described directly in web interface and in SIP tester tutorial.

Call details

The call details page is accessible from CDR screen, it contains routing/billing log (used for troubleshooting), SIP trace, CDR fields, links to download WAV recordings (see screenshot below)

CDR statistics

The web page is used to calculate CDR statistics for a specified period of time: number of minutes billed, total sales, total cost, total profit, ACD, ASR, statistics about actual number of channels

Destination set details

The page is used to manage termination-side rates. You can download rate sheet as CSV, upload from CSV, apply filter to displayed rates, make bulk modifications (add margin).

The CSV upload procedure works in this way:
  • Upload new CSV file, parse lines, for every line do this:
    • If the uploaded prefix does not exist in current list - add it into the list
    • If the uploaded prefix already exists in list - overwrite it. If the new uploaded line has non-empty "effective date" field, and the effective date is in future - keep current rate and use field "next rate" for new uploaded rate
  • Remove prefixes which do not exist in new uploaded file (in case if the supplier is no longer able to terminate the removed prefixes)
Similar web page is used to manage tariffs.

Rates statistics

General LCR view of destination sets and tariffs, used to highlight destinations with best prices for LCR routing

System status and maintenance

On this page you can check status of CPU, memory, monitor system overloads (large delays in threads), restart the softswitch, update software to latest version

Settings

The page contains general softswitch settings with descriptions; originators and suppliers self-care web portal settings; PayPal API settings for automated settings from VoIP retail users

Default call processing steps

Call flow in the StarTrinity VoIP softswitch is flexible. Here are default call processing steps:
  • Receive call
  • Authenticate call: identify originator based on IP address, prefix, CLI, PIN code, etc
    • Optionally play IVR menu for calling card user - ask PIN code, destination number
    • Determine root customer or root parent for the current originator
  • Apply trunk capacity rules of the originator (e.g. max. concurrent calls)
  • Extract dialed number (CLD) and caller ID (CLI) from INVITE packet. Apply number translation rules of originator
  • Determine tariff: take it from the root customer's service plan
  • Determine rate for the call to charge the originator: find it by the longest matching prefix with called number
  • Authorize call according to billing rules (e.g. positive balance for prepaid, or debt less than max allowed credit)
  • Determine routing group for the call. Take it from settings of the root customer
  • Select all destination sets in the routing group which have at least one route matching CLD. From the selected destination sets select route with the longest matching prefix and associated terminator. Example: for CLD = 1234567890 and two destination sets having routes with prefixes 1, 12 and 123, 1234 respectively, the software will select routes with prefixes 12 and 1234. Note: unlike with rates, where only one best matching rate has to be selected, the software selects several routes for the same destination number, but linked to the different terminators.
  • Now we the system has a list of terminators for dialing. Apply additional billing rules (e.g. positive margin), implement routing policy of the routing group
  • Apply terminator trunk capacity rules (e.g. max. concurrent calls, min. interval between calls)
  • Proxy the call to the terminators, try them one-by-one. Apply number translation rules of the terminators
  • Abort the call if billed duration goes over originator's threshold
  • Apply billing rules: decrease balance of originator based on billed duration and rate
  • Measure audio quality of the call, save results to CDR, calculate statistics, send alerts/reports (if configured)
The call processing logic can be customized by modifying CallXML script - for all or single originator/terminator. For example you can send request to a database and route the call according to result.

Installation, various use cases

The most common use cases of softswitch are retail, wholesale and termination VoIP business. In this chapter we describe how to install the softswitch for the listed use cases.
  • Retail VoIP origination
  • Wholesale VoIP business
  • GSM termination: it is cheaper to install the softswitch near GSM gateways on a windows PC, in this way saving money needed for server hosting. If you want to use the softswitch for both GSM termination and VoIP wholesale, then it is better to install it on a server in datacenter, to avoid wholesale interconnection service outages when you have problems with internet or electricity near GSM gateways.
  • IVR: IPRN, conference server
We offer a free hosted softswitch (installed on our dedicated server) here, for free if you take less than 1000 channels.

Setting up the VoIP softswitch for the first call - basic steps

  • Setup termination (supplier) side
    • Suppliers -> Add new supplier (company which is responsible for VoIP termination)
      -> Add new terminator into previously created supplier (a way to connect to the supplier, SIP trunk). Enter SIP connection parameters.
    • Destination sets -> Add new destination set (tariff for supplier, list of available country codes with prices, routes)
      -> Add new route into the new destination set or import routes from supplier's pricelist file
    • Routing groups -> Add new routing group -> Add new item that links the routing group with previously created terminator and destination set
  • Setup origination (customer) side
    • Tariffs -> Add new tariff (list of country codes and prices to charge clients) ->Add new rate into the new tariff
    • Service plans -> Add new service plan (service options like billing cycle), select previously created tariff
    • Originators -> Add new originator (company or user which sends VoIP traffic to you), select service plan and routing group. Enter SIP connection parameters
  • Make the first test call from the originator
    • CDRs -> look at the call history, see if the new call is there. Check status field
    • If there is a problem, go to CDRs -> "..." (details) -> SIP trace. Also check system log

Setting up connection to GoIP GSM gateway in single server mode

  • Install the softswitch on hosted windows server or in office on a windows PC, configure network router for SIP and RTP traffic
  • Create terminator in the softswitch:
    set up conection from GoIP GSM gateway to softswitch
  • Configure GoIP gateway:
    • Configurations - Basic VoIP:
      set up conection from GoIP GSM gateway to softswitch
    • Set up limits of using SIM cards, configure human behaviour simulation to avoid SIM blocking
  • Check status of terminator in the softswitch, it must be "registered"
  • Make a test call directly from softswitch "terminator details" web page, check details in CDR
  • Create originators (clients), service plans, tariffs, destination sets, routing groups, tariffs
  • Create a test originator for softphone, make test call via softphone. See details in CDR and in CDR - call details - routing/billing log

Web API methods

The software has a HTTP-based application programming interface (API). Main methods are described here:
  • GET /API/MainViewModel/Originators/CreateOriginator?originatorId=xxx& parentOriginatorId=xxx&maxChannelsCount=xxx&webUiUserId=xxx&webUiPassword=xxx&authUser=xxx&authPassword=xxx& balance=xxx&maxCredit=xxx&customCallerId=xxx&email=xxx&address=xxx&customerName=xxx - creates originator, is used for new user sign up at VoIP originator's website
  • GET /API/MainViewModel/Originators/GetBalance?originatorId=xxx - returns originator's balance
  • GET /API/MainViewModel/Originators/AddBalance?originatorId=xxx&value=10 - add some value to originator's balance
  • GET /API/MainViewModel/CreateCall?url=http%3A%2F%2Fyour_server%2Fget_script.aspx&user=usr&password=pass - downloads CallXML script from your web server with specified auth. user and password, creates an outgoing call with the downloaded script
  • POST /API/MainViewModel/CreateCall_Post - creates an outgoing call with the uploaded script. Returns Call-ID SIP header of the created SIP call in JSON format: {'status': 'OK', 'sipCallId': 'the_new_call_id'}. CURL example: curl http://X.X.X.X:19019/API/MainViewModel/CreateCall_Post -uadmin:admin --digest -X POST -d "@my_file.xml" -H "Content-Type:text/plain;charset=UTF-8"
  • GET /API/MainViewModel/CreateSingleCallCommand - creates an outgoing call using currently pre-configured script
  • GET /API/MainViewModel/CurrentCallExists?callerId=XXX&calledId=YYY - checks existence of current call, returns 'true' or 'false'
  • GET /API/MainViewModel/DestroyCall?sipCallId=XXX - destroys current SIP call by SIP Call-ID header
  • GET /API/MainViewModel/GetCurrentCallsCount?optionalCallXmlVariable1=XXX&optionalCallXmlVariable2=YYY - returns number of current calls, optionally filtered by CallXML variables
  • GET /API/MainViewModel/GetVersion - returns software version information - compilation time (UTC)
  • GET /API/MainViewModel/SetCallXmlVariables?callerId=XXX&calledId=YYY&direction=ZZZ&var1=value1&var2=value2 - sets CallXML variables for current SIP call, returns 'true' if call was found or 'false' if call was not found. The direction parameter of URL query is optional, it can be "in" or "out"
  • GET /API/MainViewModel/Cdr/GetCallsJSON[?sipCallId=XXX] - returns all calls from CDR memory in JSON format. Please consider setting "MaxMemoryCallsCount". The request may contain parameter "sipCallId" if you need to get information about specific call
The API methods by default require authentication - you need to use same credentials as in web UI. To turn off authentication, you can add your IP address into whitelist using setting "WebApiTrustedIpAddresses".
There are more undocumented methods available, you can see them in Fiddler when using the web interface. Please let us know if you need some new API method
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