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 call charges. In some respects, VoIP peering is really just an extension on "in-network" IP-routing, which is usually free.
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 dialling instead of direct DID (Direct Inbound Dialling) number dialling. Keeping in mind that SIP URI dialling will typically follow the format of something like: firstname.lastname@example.org, or email@example.com, 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 dialling 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 Voipfone account and connected ATA, IP-Phone, or softphone, I dial the Sip Broker access code as enabled by Voipfone: *1.
- 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 Voipfone connected ATA, IP-Phone, or Softphone I dial:
- *1 (access to Sip Broker network via Voipfone)
- *011 (Sip Broker's own SIP Code)
- 188888 (Sip Broker Welcome Message number)
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 SPA112 ATA to include the following dial plan addition: |*1*x.|
Dialling 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. firstname.lastname@example.org.
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:
*email@example.com - Sipbroker Welcome Message
Whether the URI will 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 dialling, 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 Voipfone account using the 3CXPhone softphone.
Why You Should Care About VoIP Peering
VoIP peering can save you a substantial amount of 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 call charges.
For example, perhaps you have family, friends, or know of businesses who have VoIP services 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 a Sip Broker network as I earlier explained in this thread.