Antisip - SIP Services

FAQ

If you don't find your answers, contact me by email.
contact details from here or in page's footer!.

General questions
sip.antisip.com is a SIP services. After you have created an account with us, you can use a VoIP Application to make SIP (VoIP) calls. The objective is to provide a high quality service with minimal server features. In fact, we provide registration, routing and voicemail. With SIP, other services/features (presence, messaging, etc...) are provided by end user applications!
Understand that the service is feature free and provides 2 kinds of access, usually not compatible together... You can use our service for basic SIP calls, but you can also use the service for webrtc like calls. The requirements are usually different and usually, the setup are NOT interoperable...
sip.antisip.com
Visit our registration webpage at www.antisip.com Note that we do ONLY accept lowercase username, and do not accept any more complex UTF8 username. This was too difficult for users to configure their app correctly. In your SIP application, you MUST use a lowercase username and you MUST use a lowercase authentication-user (which is your account username in lowercase)
SIP is a standard protocol and you will find many applications or devices compatible with SIP. Some are good, others are... usable! You can use any SIP Application to connect to our sip.antisip.com service. If you own an Android device, we advise you to use VoIP By Antisip, our Android app from Google Play.

Get it on Google Play
Get the voip by antisip Android app from Google Play.

We also offer websocket access and provide a web page to make webrtc calls through our services.. Connect on the web page with your username and password, and go to the room page. Send a link to a friend (no account required), or call a contact from antisip (account required). THE WEBRTC SERVICE CURRENTLY WORKS ON CHROME, MAY BE OTHER AS WELL IN THE FUTURE... TEST YOURS...
You need to first download a SIP Application. If you own an Android device, prefer VoIP by Antisip. Then, configure:
  • domain: sip.antisip.com
  • username: -your account name at sip.antisip.com-
  • password: -your account password at sip.antisip.com-

If you need to use Voip By Antisip for webrtc call, you also need to configure those settings:
  • stun: stun.antisip.com
  • RTP profile: UDP/TLS/RTP/SAVPF

With the above settings, Voip By Antisip can make calls to the room page
Many NAT box are broken, many network are blocking VoIP, etc... Many *bad* reasons for a SIP application to just have no answer from a SIP proxy. To workaround, you should configure with those optional settings. This will help to hide the traffic...
  • protocol: TLS (+Verify certificates)
  • outbound proxy: sip.antisip.com:9091
NO... It looks people using sip.antisip.com are sometimes expecting to be able to call PSTN, traditional numbers. This is NOT possible. We are NOT providing gateway services. This services is free to use and PSTN access requires money. So if you are looking for such gateway services, this is not the right place! To be clear, we provide only VoIP to VoIP services.
Your username must be in lower-case... This is a rule at sip.antisip.com. The username (or authentication-user user) MUST be in lower case.
VoIP By Antisip
If you own an Android device, we advise you to use VoIP By Antisip, our Android app from Google Play.

Get it on Google Play
Get the voip by antisip Android app from Google Play.

You can follow the magic wizard which can help you to create a new account, or configure an existing account.

You can also enter the settings, and fill only the 3 required SIP configuration parameters:

  • domain: sip.antisip.com
  • username: -your account name at sip.antisip.com-
  • password: -your account password at sip.antisip.com-

If you need to use Voip By Antisip for webrtc call, you also need to configure those settings:
  • stun: stun.antisip.com
  • RTP profile: UDP/TLS/RTP/SAVPF

With the above settings, Voip By Antisip can make calls to the room page
If you need basic help with sip.antisip.com and Voip By Antisip, then, please visit our html tutotial. You'll find guidelines and snapshots
This is a tutorial to help you with Voip By Antisip. It will provide guidelines and all steps required to create an account at https://sip.antisip.com.

Watch Tuto 1!

This is a tutorial to help you with Voip By Antisip. It will help you to fill your Address Book with SIP friends to simplify Voip By Antisip usage.

Watch Tuto 2!

In order to have SIP access, you should create an account and configure basic settings. The basic settings will let you interoperate with most SIP applications.
  • SIP domain: sip.antisip.com
  • SIP username: your-username
  • SIP password: your-password

Optionaly, this may help to bypass network restrictions and workarounds bad networks

  • SIP protocol: TLS (+Verify certificates)
  • Outbound proxy (if troubles): sip.antisip.com:9091 (for TLS)

If you need to use Voip By Antisip for webrtc call, you also need to configure those settings. If you set the value below, you will most probably loose interoperability with more basic SIP phone...
  • stun: stun.antisip.com
  • RTP profile: UDP/TLS/RTP/SAVPF

With the above settings, Voip By Antisip can make calls to the room page
In order to benefit from our service, you need to configure your app with a list of mandatory settings. Without them, you won't get security, privacy and you may not get audio or video as well... Also understand that your correspondant must be compatible with those settings.
  • TURN server/username/password: Get a list!
  • STUN server: Get a list! or stun.antisip.com
  • RTP Profile: UDP/TLS/RTP/SAVPF
  • Protocol: TLS (+Verify certificates)
  • Dual IPv4/IPv6: enabled
  • Outbound proxy (if troubles): sip.antisip.com:9091 (for TLS)

Antisip SARL.
118, Rue Saint Georges, 69005 LYON
FRANCE
Aymeric Moizard
amoizard_at_gmail.com
P: +33 6 32 51 25 86