Outbound SIP

Introduction

The Ziron Outbound SIP service allows you to terminate voice traffic from your SIP platform or switch. It is generally not suitable for directly providing service to end users. Whilst this article focusses mainly on the technical implementation of an interconnect for outbound voice traffic with Ziron, you should also familiarise yourself with the Acceptable Use Policies for the relevant country/countries.

 

Hostnames

By default you should use the hostname sip.ziron.net. This will resolve to the closest Ziron proxy. You should not attempt to send traffic directly to the IP addresses associated with this hostname as these may change without warning. 

Where possible, your SIP software should support SRV records. 

If you wish to specify a particular Ziron proxy location, you can do so by adding the location to the hostname as follows:

  • London: sip.lon.ziron.net

Calls should be sent in E164 format, e.g. 442079460123@sip.ziron.net. Please note that whilst other number formats may result in successful call routing, this is on a best efforts basis and is not guaranteed. 

 

Authentication

We currently support two methods of authentication:

IP Address authentication

Outbound calls are authenticated by the originating IP only. This should only be used where your SIP server, PBX or soft switch has a static IP address. 

Username/Password authentication

Where you cannot guarantee the IP address of your equipment remains static, or you wish to support multiple SIP accounts or user accounts on the same IP address, you can use this authentication method. 

 

Caller ID

Valid CLI must be presented on all calls on E164 format (e.g. 442079460123). Failure to provide this information may result in your call being rejected without warning.

If you wish to 'withhold' the calling number (i.e. not display to the end user), you should set the 'From:' header to 'Anonymous' and ensure that a valid CLI is given in the P-Asserted-Identity header, for example:

From: "Anonymous" <sip:anonymous@anonymous.invalid>
P-Asserted-Identity: "442079460123" <sip:442079460123@example.com>
Privacy: id

Or for systems that do not support P-Asserted-Identity, you may use the Remote-Party-ID header, e.g.:

From: "Anonymous" <sip:anonymous@anonymous.invalid>
Remote-Party-ID: <sip:442079460123@example.com>;privacy=full;screen=yes;party=calling;

 

DTMF

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

 

Codecs

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): 

  • 185.43.128.6 - London
  • 185.43.128.7 - London

RTP Media (udp ports 1024-65534):

  • 185.43.128.0/26

 

 

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