SipWarp

Connect an account

Press Account at the bottom of the window. Fill in the server, username, and password from your provider, then Save account. The status line should say Online. If it stays on Registering, the server name or password is wrong.

Transport is udp unless the provider told you to use TLS. Port 0 means SipWarp tries 5060, which is what most of these providers expect.

For a US or Canada call, dial 1, then the area code and number. Example: 13125551212. A 10-digit number often fails on voip.ms.

voip.ms

  1. In the voip.ms portal, create a sub-account. The username looks like 123456_home. That full name is the username, not just the part after the underscore.
  2. Choose a server close to you, such as chicago1.voip.ms or newyork1.voip.ms. The portal lists them.
  3. In SipWarp, set Server and Domain to that server. Username and password are the sub-account. Display name is your name.
  4. In Settings, Network, set RTP min 10001 and RTP max 20000. Turn STUN on and set the server to stun.voip.ms.
  5. Codecs: leave G.711u first. That is what voip.ms uses toward a cell phone.

Caller ID is set in the voip.ms portal (Account settings or the sub-account). SipWarp can also send an outbound caller ID. The portal will ignore it unless that sub-account is set to pass caller ID from the device, and the number has to be one voip.ms already allows.

Twilio

  1. In the Twilio console, create a SIP domain, such as yourname.sip.twilio.com.
  2. Create a credential list (a username and password) and attach it to that domain so the phone is allowed to register.
  3. Server and Domain are the SIP domain. Username and password are the credential list.
  4. Buy a number and point the domain at a voice webhook, or outbound calls register and then fail. Twilio's own SIP domain page shows that step.

Telnyx

  1. Create a credential connection. Telnyx generates the username and password. Choose Credentials, not IP authentication, for a home phone whose address changes.
  2. Server and Domain: sip.telnyx.com. Port 5060, transport udp. (Use 5061 and tls only if you turned on encryption.)
  3. Attach an outbound voice profile, and a phone number if you want people to call you.

Anveo

Anveo's SIP server is sip.anveo.com. The username and password are the SIP device in your Anveo account. Anveo's usual SIP port is 5010, not 5060. Put 5010 in Port. If the portal shows a different server or port, use that.

IPComms

IPComms sends the SIP server, username, and password when the trunk is created. Put that server in Server and Domain. Use their username and password. Leave Port at 0 unless their email names a different port. UDP is the normal transport.

RTP ports and STUN are the next page if the account is Online but the call has no audio.