Monday, 5 November 2012

Does Your VSP Allow VoIP Peering?

VoIP peering is essentially where two VoIP service providers (VSP) have a "bilateral" agreement in place that allows VoIP calls to be routed between their respective networks.

Because VoIP peering agreements typically allow for direct IP inter-connectivity between two or more networks, all calls between the peering networks can circumvent the PSTN networks (public switched telephone network) which in turn enables extremely low connectivity costs for users that would otherwise have to pay substantial PSTN toll charges.  In some respects, VoIP peering is really just an extension on "in-network" IP-routing, which is usually free, or extremely cheap (small fractions of a penny/minute).

When multiple VSP's agree to share traffic between their respective networks, they create what is known as a "federation" of peers. Through federation, the VSP's are able to grow their geographic areas of coverage at a substantial savings over PSTN call routing. And, hopefully these savings are passed onto their customers.

Typically, in order to directly call a user on another VoIP network that is peered with your provider's network, you will use SIP URI dialing instead of direct DID (Direct Inbound Dialing) number dialing.  Keeping in mind that SIP URI dialing will typically follow the format of something like:, or, etc.  Now, when you dial using a SIP URI, the call will be IP routed over the Internet or other IP backbone between the peering networks.

In some cases, you may be able to call another person on a peered network by directly dialing their DID or network account number.  But, this may well vary form VSP to VSP and the details of their peering agreements.  Otherwise, URI dialing will typically be the most reliable method of calling between peered networks.

Using Sipbroker For Peer Access
Some VSP's will allow direct access to a VoIP Peeing gateway known as "SIP Broker".  Sip Broker acts as a free central gateway for peering between a multitude of world-wide VoIP services who have peered with Sip Broker.  Anyone who dials into the Sip Broker network just needs to know the "SIP code" of any other peered network they want to call into, for free.

The following are some examples of how I call into networks peered via Sip Broker:
  • Using my or CallCentric account and connected ATA, IP-Phone, or softphone, I dial the Sip Broker access code as enabled by CallCentric:  **275.  
  • The access code will then be followed by the "Sip Code" that has been assigned to the VoIP provider whose network you are trying to call into.  The Sip Code will consist of a 3 to 5 digit code preceded by the asterisk (*).  i.e *xxx, *xxxx, or *xxxxx.
  • Following the SIP Code, you will dial the DID, account number, or other assigned number of the contact you are trying to connect to.  The length of the destination number depends on the peer network you will be calling to and may be of variable lengths.

Calling SIP Broker Welcome Mesage:

From my or CallCentric connected ATA, IP-Phone, or Softphone I dial:
  • **275 (access to Sip Broker network via CallCentric)
  • *011  (Sip Broker's own SIP Code)
  • 188888  (Sip Broker Welcome Message number)
The above dialed sequence of codes are concatenated (added together) as a single dial string:

Because of the special dial string required to call into Sip Broker, it may be necessary to edit your ATA or IP-Phone's "Dial Plan".  For example, I had to edit my SPA2102 ATA to include the following dial plan addition:  |**275*x.|

Dialing Into Peered Networks Using SIP URI's
SIP URI's (Uniform Resource Identifier) allow calls to be routed directly over the Internet.  URI's follow a format similar to email URL's (Uniform Resource Locator).  i.e.

If you have a VoIP ATA, IP-Phone, or Softphone that is enabled to dial SIP URI's directly, you can try calling the following test VoIP URI's:

* - Sipbroker Welcome Message  - Current Time Announcement (EST) - Welcome Message at - Zipdx wideband demo recording - VoIP Users Conference (Listen-in on Fridays 12 noon EST)

Whether these URI's work or not will depend on various factors, such as the user agent you are calling from (ATA, IP-Phone, or Softphone), your VSP (VoIP Service Provider), whether your VSP allows direct URI dialing, and if your VSP has peering agreements with any of the URI domains you make URI calls to.

I've tested all the above SIP URI's from my CallCentric account using my 3CXPhone softphone.

If you use, you will need to first add the above SIP URI's to your "Phone Book" via your online dashboard, and store them as *75xx phonebook quick dial numbers. Once stored into the Phone Book, all you have to do from the 3CXPhone is to dial the associated *75xx phone book quick dial code and the SIP URI number will now connect.

Why Do You Care About VoIP Peering?
VoIP peering can save you substantial money on some of  your VoIP calls by bypassing the telco's PSTN call routing.  It will allow direct call routing between VoIP network services without incurring PSTN toll charges.

For example, perhaps you have family, friends, or know of businesses who have VoIP service through a different VSP than you do.  Well, if your VSP and theirs have peering agreements, you may be able to dial them direct via their SIP URI... for free.  Or, if not direct, you may still be able to call them via network as I earlier explained in this article.

Happy peering!