MicroSIP datasheet

MicroSIP is a SIP softphone based on the PJSIP stack available for Microsoft Windows. It facilitates high quality VoIP calls based on the open SIP protocol.

IP Ports and Protocols used by SIP Devices. The following ports are associated with a successful communication:

IP Ports and Protocols used by SIP Devices
Trust Over IP	Port	Type	Description
SIP Server Ex. 192.168.1.1	5060	UDP	SIP endpoints
SIP Server Ex. 192.168.1.1	10000:65000	UDP	RTP/SRTP Media

A security group controls the traffic that is allowed to our IP; reach and leave the resources that it is associated with.

General

Softphone usage modes

Single call mode - single window, basic functionality. Enabled by default.

Extended mode - two windows, multiple calls, conferences, attended transfers.

Communication types

Calls through SIP server / PBX - select "Add Account" after installing.

Direct calls by IP address (or domain name). Works out of the box, using the "Local Account".

After automatic startup or when you close the main window MicroSIP will be minimized to the system tray.

MicroSIP does not require the installation of additional libraries, runtimes or frameworks.

Dialpad

Mainly used for dialing or sending dual tones (DTMF). Various input formats are supported.

Examples

1-800-567-46-57

1234

1234@sip.server.com

1234@sip.server.com:5043

192.168.0.55

Or even complete SIP URI with optional microsip extensions

"Name" <sip:extension@sip.microsip.org;parameter1=xxx?Custom-Header=yyy>,dtmf_sequence

There is sound quality indication

Control switches and buttons

DND (switch) - Do not disturb mode

FWD (switch) - Automatic forwarding of incoming calls. Set up in the settings

AC (switch) - Automatic conference for incoming calls after answering a call

AA (switch) - Automatic answer. Set up in the settings

CONF (button) - Invite a participant to a conference call

REC (button) - Current call recording. Set up in the settings

Account

SIP server

Your account SIP server.

SIP proxy

Your account SIP proxy or a chain of proxies.

Examples: 192.168.1.1, 192.168.1.1:5070, 192.168.1.1 192.168.15.1, 192.168.1.1;hide, ";hide" parameter can solve impossibility of registration or calls due to server configuration.

Username

Your account username.

Domain

Your account domain.

Login

Username for authentication. If empty, will be used Username.

Password

Your account password.

Display name

Your name, remote party will see it in incoming calls and messages.

Dialing Prefix

International calling prefix for numbers in local format (must begin with "+" or "00"); or a simple prefix for each dialing phone number.

Dial Plan

Transforms dialing number according to pattern. Numbers that do not match any patterns are blocked. Patterns are separated by a pipe symbol: |. The entire value can be enclosed in brackets ().

x	"x" represents any character
[sequence]	Enter characters within square brackets to create a list of accepted digits. Numeric range: enter [2-9] to allow the user to enter any one digit from 2 through 9. Numeric range with other characters: enter [16-9*] to allow the user to enter 1, 6, 7, 8, 9, or *.
<dialed:substituted>	Replaces one sequence with another. Or inserts some sequence inside a number: <:substituted> Example 1 : <8:1555>xxxxxxx If user dials 81234567, the system transmits 15551234567. Example 2 : <:1>xxxxxxxxxx If user dials 1234567890, the system transmits 11234567890.
. (period symbol)	Represents zero or more entries of the previous digit. Example, 01. => 0, 01, 011, 0111, ...; x. => matches any dialed number.

Example: Replace + with 00, allow any other numbers.

<+:00>x.|x.

Complex rule example:

[3469]11|0|00|1[2-9]xx[2-9]xxxxxx|<:1>[2-9]xx[2-9]xxxxxx|<:1618>[2-9]xxxxxx|<:1618555>6[2-4]xx

Hide Caller ID

Your PBX must allow this feature. Otherwise, you probably won't be able to call. For FritzBox, clear this check box and change Display Name to "Anonymous".

Voicemail Number

Voicemail access number. If empty, microsip will try to determine it automatically.

Media encryption

Disabled - never use encryption, Optional - use encryption when remote party supports encryption, Mandatory - use encryption always. Recommend value: Optional.

Transport

The value depends on the configuration of your SIP server. Failsafe value: UDP. Best value: TLS. TCP is good, but is may not work with your router/NAT due to SIP ALG enabled. "UDP+TCP" is a mix of UDP (for small request) and TCP (for large).

Public address

Can be used to solve call flow and media delivery issues when you don't have dedicated public IP address. You can manually specify IP address or hostname for Via, Contact and SDP. It can point to one of the interface address OR it can point to the public address of a NAT router where port mappings have been configured. For automatic public address detection and rewrite you can use Allow IP rewrite feature or use STUN server.

Local port

By default MicroSIP tries to listen on standard SIP port - 5060. If port is busy by other application, MicroSIP will listen on random port. You can manualy change port to any.

Publish presence

Sends on SIP server publish query, it means that other subscribed contacts can see your status and can pickup your incoming calls (BLF functionality). Besides, often you must specify which contacts have right to see your presence information - you can done this for example via SIP provider webpage. Your SIP server must support this feature.

ICE

Helps to find shortest way for media streams and reduce media latency. It is usefull when possible direct P2P connection without SIP provider mediagate. Enabling ICE can cause problems with in media delivery if SIP server configured incorrecly.

Allow IP rewrite

Can be used to solve call flow and media delivery issues when you don't have dedicated public IP address. If enabled, MicroSIP will keep track of the public IP address from the response of REGISTER request. Public IP will be used in later SIP queries in Via, Contact and SDP. See also: Public address, STUN.

Disable Session Timers

Specify the usage of Session Timers. Try to disable Session Timers if your calls dropps after XX minutes. Recommended value: unchecked.