Microsip

From VoIP.ms Wiki

[draft revision]

[checked revision]

Revision as of 01:53, 26 July 2023 (view source) Roy (Talk | contribs) ← Older edit		Latest revision as of 15:06, 28 June 2025 (view source) Roy (Talk | contribs) (→Basic setup)
(17 intermediate revisions not shown)
Line 12:		Line 12:
	[[File:Microsip.png|300px|border]]		[[File:Microsip.png|300px|border]]
		+	<br>
		+	<br>
	3. Fill in the following		3. Fill in the following
Line 25:		Line 27:
	Finally, press '''Save''' and you are done configuring MicroSIP.		Finally, press '''Save''' and you are done configuring MicroSIP.
		+
		+	<div style="margin-left:10px;
		+	padding:5px;
		+	border-left: 6px solid rgb(208, 56, 45, 0.8);
		+	box-shadow: 8px 6px 9px -2px #f2f2f2;
		+	background-color: rgba(208, 56, 45, 0.08);
		+	width:70vw;">
		+	🚨 '''IMPORTANT'''
		+	: <b style="color:#b30000;">Are you located outside of the Americas?</b> If so, please use any of the international servers [https://wiki.voip.ms/article/Choosing_Server Servers list]
		+	: US/CAN servers might not be reachable from some countries outside the Americas, international servers can be reached from anywhere in the world.
		+	</div>
	[[File:Microsip3.png|300px|border]] [[File:Popserver.png|600px|border]]		[[File:Microsip3.png|300px|border]] [[File:Popserver.png|600px|border]]
		+	<br>
		+	<br>
	4. Go to the dialpad and test the audio dialing 4443		4. Go to the dialpad and test the audio dialing 4443
	[[File:Microsip4.png|300px|border]] [[File:Microsip5.png|300px|border]]		[[File:Microsip4.png|300px|border]] [[File:Microsip5.png|300px|border]]
		+	<br>
		+	<br>
		+	5. Finally, define a caller ID for your username so you can start making outgoing calls:
	<br>		<br>
	<br>		<br>
		+	<b>For the main username: </b><b style=color:#b30000>Go to Main Menu -> Account settings -> General tab and fill the "Caller ID number"</b><br>
		+	<b>For a subaccount/extension: </b><b style=color:#b30000>Go to Subaccounts -> Manage subaccount -> Edit subaccount and fill the "Caller ID number"</b><br>
		+
		+	In both cases, make sure to wait 3 minutes after setting the caller ID number and trying to dial out <br>
		+
		+	[[File:CallerIDnumber.png|500px|border]]
		+
		+	==Additional Settings (Optional)==
		+
		+
		+	===TLS SRTP===
		+
		+	In order to enable TLS SRTP (call encryption), you will first need to enable this feature by modifying your main account under Main Menu > Account Settings > Advanced tab and setting '''Encrypted SIP Traffic''' to '''Yes'''
		+
		+	Once done, go into the menu option (arrow pointing down) on the top right of MicroSIP > Edit Account and set the following options:
		+	:- Media Encryption: Mandatory SRTP
		+	:- Transport: TLS
		+
		+	[[File:microSIPTLS.png|thumb|none|600px|Click to enlarge]]
		+
		+	Save and then head once again on the top right of MicroSIP > Settings and once there, modify '''Source Port''' option to 5061 and RTP to 10001 to 20000.
		+
		+	Save these settings and you will now
		+
		+	[[File:microSIP4.png|thumb|none|600px|Click to enlarge]]
		+
		+
		+	===Some additional settings===
		+	By heading into the menu option (arrow pointing down) on the top right of MicroSIP > Settings there are some settings that can be improved in order to avoid any issues while on a call:
		+
		+	:- Enabled Codecs: Only have G711u allowed (or GSM if you have a slow internet connection)
		+	:- Source port: Set it to 5060 (5080 or 42872 for alternate ports) or 5061 (5081 or 42873 for alternate ports) for call encryption
		+	:- RTP Ports: Set the minimum to 10001 and maximum to 20000
		+	:- EC: Make sure this option is enabled. This is an Echo Cancellation option
		+
		+	Otherwise, there are many other options that you can customize like the ring tone you will hear when someone calls among many other options you can customize or enable that you can have a look into. You can also consult [https://www.microsip.org/help/this link] if you have doubts on the options MicroSIP offers.

---

Latest revision as of 15:06, 28 June 2025

Contents 1 Tutorial Video 2 Basic setup 3 Additional Settings (Optional) 3.1 TLS SRTP 3.2 Some additional settings

Tutorial Video

This is the very basic set up for MicroSIP, there are only a few steps needed to have your device up and running. If you wish to improve your softphone performance or enable some features, refer to the advanced section.

Basic setup

1. Download and install MicroSIP from this link.

2. Once downloaded and installed, start MicroSIP and on the welcome screen head onto the arrow pointing down on the top right of the softphone application > Add Account.

3. Fill in the following

Account name: Type your phone number or write any name, it's just an identifier
SIP Server: Type the server of your number, check it out from your portal, i.e. denver1.voip.ms
Username: Type your account number or subaccount name
Domain: Type the server of your number, check it out from your portal, i.e. denver1.voip.ms
Password: Type your password
Display name: Type your name or company name (NOT guaranteed to be displayed)

Finally, press Save and you are done configuring MicroSIP.

4. Go to the dialpad and test the audio dialing 4443

5. Finally, define a caller ID for your username so you can start making outgoing calls:

For the main username: Go to Main Menu -> Account settings -> General tab and fill the "Caller ID number"
For a subaccount/extension: Go to Subaccounts -> Manage subaccount -> Edit subaccount and fill the "Caller ID number"

In both cases, make sure to wait 3 minutes after setting the caller ID number and trying to dial out

Additional Settings (Optional)

TLS SRTP

In order to enable TLS SRTP (call encryption), you will first need to enable this feature by modifying your main account under Main Menu > Account Settings > Advanced tab and setting Encrypted SIP Traffic to Yes

Once done, go into the menu option (arrow pointing down) on the top right of MicroSIP > Edit Account and set the following options:

- Media Encryption: Mandatory SRTP

- Transport: TLS

Save and then head once again on the top right of MicroSIP > Settings and once there, modify Source Port option to 5061 and RTP to 10001 to 20000.

Save these settings and you will now

Some additional settings

By heading into the menu option (arrow pointing down) on the top right of MicroSIP > Settings there are some settings that can be improved in order to avoid any issues while on a call:

- Enabled Codecs: Only have G711u allowed (or GSM if you have a slow internet connection)

- Source port: Set it to 5060 (5080 or 42872 for alternate ports) or 5061 (5081 or 42873 for alternate ports) for call encryption

- RTP Ports: Set the minimum to 10001 and maximum to 20000

- EC: Make sure this option is enabled. This is an Echo Cancellation option

Otherwise, there are many other options that you can customize like the ring tone you will hear when someone calls among many other options you can customize or enable that you can have a look into. You can also consult link if you have doubts on the options MicroSIP offers.