SIP Device Configuration

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Most all SIP devices follow the RFC standards which makes it a bit easier to get up and running.
Here are the basics to keep in mind when connecting your SIP devices and ATA’s.
 

SIP Proxy vs Realm/Domain:

We use a “Realm” or “Domain” for authentication purposes which is unique to your account.
This must be sent with SIP Registrations and Invites. Example; ‘mycompany.nurango.io’

The SIP Proxy is the actual SIP server that you want to connect to. You should have received that info via email.

Codecs:
We accept the following codecs;

Audio

  • G.722 – Wideband (HD audio)
  • G.711 – uLaw (North American standard) 
  • G.711 – aLaw (European standard)
  • Speex
  • GSM (but why)

Video

  • VP8
  • H264
  • H263
  • H261 

MWI (Message Wait Indicator)

Used for voicemail notifications. You will receive a notice and usually a flashing light when new voicemail is received.
This feature needs to be enabled in your phone/device settings.

Presence
If presence is turned on and enabled in your SIP device, it will update our server with your status. For example, when you turn your status to “Busy” in a softphone, we will know you are unavailable to take calls. Can also be used with various call center applications.

Security
We accept and endorse SIP/TLS and SRTP! Most IP handsets, devices and softphones support this feature now-adays.
Please refer to your device manual for your specifics.

  • Port: 5061 (TCP)
  • TLS Cipher: TLSv1
  • SRTP Cipher (SDES): AES_CM_128_HMAC_SHA1_80
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