Configuring Generic SIP Devices | nurango academy

Configuring Generic SIP Devices

The following are settings that our platform supports and can be used generally with any SIP device.

 

Transport Methods:

  • SIP: TLS (encrypted), UDP, TCP
  • Video: H.264, H.263, VP8
  • Fax: T.38
  • Audio: SRTP, UDP
  • Other: WebRTC/TLS

Lookup Methods:

  • "A-Record" with Load Balancing
  • SRV
  • Direct via IP

Ports:

  • TLS - 5061
  • UDP - 5060
  • TCP - 5060
  • WebRTC - 5065

Codecs:

  • G.722 HD
  • G.711 u/a

DTMF:

  • RFC2833
  • Inband

 

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

Firewall Considerations:
If you need to Whitelist our IP's on your firewall, please open a ticket to get the current list.

Here's what you will need to know;

Outbound Proxies: Used for Registrations and Outbound calling.
Inbound Nodes: IP's used for inbound calls.
DNS is available for both groups to make things easier if your device supports DNS to IP resolving.

NAT and Audio Considerations:
Although some devices will try and enforce port ranges on audio, this is often the biggest challenge and reason for One-way or No audio.
We recommend you Whitelist our IP's on every UDP port range to ensure successful audio delivery.

Backup and Failover Options:
We absolutely recommend customers have a SIP device that supports a Secondary Proxy or Trunk Failover.
There are a number of IP phones and IP-PBX's that do this and should be a purchasing factor if up-time is a proiority for your company.

Our side -
Dual Proxy Nodes: We have more than 1 registration node and if one becomes unavailable you can failover to the secondary one.

Your side - 
Failover: In the event that your equipment becomes unavailable you can failover your calls to either a SIP URI or an external phone number.

Other Considerations:
It's importand to note the difference between SIP Proxy and SIP Realm (sometimes referred to as "Domain").
The SIP Proxy is where your actual calls and registrations are sent to.
The Realm/Domain is sent in the SIP packets as part of the Authentication process. Think of it as a second authentication method in addition to your username and password.

Generally, SIP devices will have fields to differenciate the two such as;
"Outbound Proxy" for the SIP Proxy, and "SIP Server", "Realm" or "Domain" for the latter.
Check this KB for some of the more popular device setup guides or refer to your devices manual for specifics to you.