Inbound SIP


The Ziron Inbound SIP service allows you to receive voice traffic to your SIP platform or switch. It is generally not suitable for directly providing service to end users. 



Calls will be delivered to the SIP URI set in the relevant endpoint associated with the number being called. You may use one of the following placeholders anywhere in your endpoint URI:

%e164 The called number in E164 format
%national The called number in national format

(%did is a legacy placeholder identical to %e164)

For example, a call destined for +442079460123 using the following URI:

will be expanded to:

This also allows the use of tech prefixes, e.g. to prefix all invites with 12345, you could use the following URI:

We will follow port and protocol designations, for instance:;transport=tcp 

We will also follow any preferences found in SRV records on a 'best efforts' basis.


Caller ID

CLI is provided in E164 format on a 'best efforts' translation basis. You can change this to one of the following via the Ziron dashboard or API:

format description
raw as received from the originating network
e164 e164 format
national national format for the called number country 


Please note that where CLI cannot be validated, the 'raw' data will be passed through.

CLI data is given in the P-Asserted-Identity header, and we support unknown/withheld CLI via the Privacy header.



DTMF is only supported out-of-band as per RFC2833.


Extra Headers

Calls may contain the following additional headers:

header description
X-ziron-country-iso Country ISO code of the called number, e.g. GB
X-ziron-number-type Type of the called number, e.g. local, national, mobile
X-ziron-number-sid Unique identifier representing the called number 
X-ziron-endpoint-sid Unique identifier representing the endpoint 
X-ziron-trunk-sid Unique identifier representing the trunk



We currently support the following codecs:

  • G711a /A-law
  • G711u /μ-law


Firewall / IP information

You should allow traffic to and from the following IP addresses and network ranges in both directions:

SIP signalling (udp port 5060, tcp port 5060, tcp port 5061): 

  • (legacy)
  • (legacy)

RTP Media (udp ports 1024-65534):

  • (legacy)
  • (legacy)
Have more questions? Submit a request


Article is closed for comments.
Powered by Zendesk