In regular network situations, this technology will keep work as a sip terminal. It keep detect the network connection, when the sip server is out of work, it will take up the backup mission. When the internet is broken or the sip server is down, it can detect them out. Even in the worst case, the power is lost, the PSTN intelligent routing tech will forward all out boundary calls to PSTN lines. PSTN lines will not be affected by the terminal’s power issues.

They will receive VoIP calls from providers that they have contracted with to provide traffic for a particular country or area. Typically, these providers will send the calls with a specific number format and SipTang SHANG332 needs to be configured to accept the call and terminate it to the first available analog line or E1 channel and dial out the correct digits.

When setting up for this type of application, the following issues should be taken into consideration and in some cases are necessary to know before configuring.

You must decide what is the best connection to have from the PSTN provider to SipTang SHANG332. Many things must be taken into consideration for this.

n        Price

This is always the biggest consideration. Not so much as the initial installation, but the monthly charges and rates that you get will, many times, determine the type of connection. If you choose a T1 (or E1) connection, the monthly fee will be more expensive that that of several analog lines. Depending on the amount of traffic that you will receive, you may not be able to make enough money to pay for a T1 or E1 line, or you may need to charge a higher rate to your provider and at some point the provider may decide to switch to a less expensive competitor.

n        Number of calls to support.

You need to determine how many calls you want to support and see what type of PSTN connection and how many will support your requirements. You should choose T1(or E1) connections when the traffic is more than 30 concurrent calls, otherwise, Analog lines would be better choice.

n        Availability.

Not all countries have all types of connections. Some may only have analog connections. You should check with your local PSTN provider about this.

You need to determine how much bandwidth you will need. You can do this through some simple math. You first determine how many VoIP calls you want active at the same time on your SipTang SHANG332 and multiply this by the bandwidth required based on your audio compression.

If you plan to use G.729, then figure on about 19kb per call in each direction. If you plan to use G.723.1 @ 6.3kb, the bandwidth will be about 13.5kb per call in each direction.

For the quality, this will mainly depend on the ISP that you use for your Internet connection. If it is a lower tier ISP, the quality may not be there in that you may experience high packet loss or long delays on your Internet connection. You should discuss this with your ISP.

To reduce the effect of bad internet connection, SipTang SHANG332 provide many QoS features, such as Dynamic Jitter Cancellation, Voice Activity Detection, Silence Suppression, Echo Cancellation(up to 128ms), etc.

At a minimum, you will need to know what the number/digit pattern is that your providers will send to you. Typical patterns are international prefix (like 00 or 011) + CountryCode + number, or Countrycode + number. In many cases, providers may add a special prefix to the front of the number. This is usually a 4 or 5 digit number (could be more or less) that gets added to the front of all numbers and is sent as part of the phone number. For example, if the prefix is 6789, then they may send you the number as 6789+0101234567890. It is very important to know what the digit pattern is that your provider will send to you as SipTang SHANG332 needs to have these patterns configured in it to allow the call to terminate.

You may also want or need to know the provider’s IP address for security.  SipTang SHANG332 allows you several ways restrict access to it. The easiest way is a simple access list of allowed IP addresses. More complex would be with a radius server for authentication. SipTang ZHOU500 have an integrated billing system which can be used for authentication.

You may have several units installed at the same location to terminate calls. In these situations, most customers are looking for calls to be distributed evenly over all the units or in the least, if the first unit is filled, for the call to overflow to the next. This feature can only be used from the call origination point, not from the termination. Once the termination SipTang SHANG332 receives a call from IP, it cannot re-route it back to IP to another termination unit. You should discuss this with your provider/originator to have them perform this function at their side.

In some cases, you may have several connections from different PSTN provider to avoid the effect of single PSTN lines’ failure. You can use SipTang ZHOU500 to route the traffic to different PSTN lines for load balancing, redundancy and overload protection, in order to assure the stability, robustness and scalability.