A VoIP Terminator

A VoIP terminator is one who takes VoIP calls off internet and delivers them to PSTN phones. Therefore, selection of a terminating partner that can transmit your calls to their destinations with better quality is also vitally important. While selecting a terminator, following different issues should be considered to provide better-quality service.

• Number of calls managed simultaneously by the network
• The alternate way to transfer the call to it desired destination in case of any fault/failure occurred in the network
• Supported CODECs for coding and encoding purposes
• Overall setup of the network
• The protocol used by the termination network

In telecommunications services, emerging use of VoIP (Voice over Internet Protocol) and other services (like VOD, IPTV etc.) has made QoS monitoring an essential element for high-quality service. It is used to monitor the quality of a network in terms of transmission, error rate and other characteristics that can be maintained to improve the quality. Quality of any service depends on the traffic flow as well as the network of terminating partners.

Most commonly used signaling protocols are H.323 and SIP (Session Initiation Protocol) and can be used in the same network. Both these protocols are used in VoIP (Voice over IP) and Video Conferencing. H.323 provides compatibility between VoIP equipment and equipment from different manufacturers. SIP is introduced after H.323 but is now much popular for VoIP ervices. It is specifically designed to attain simplicity and scalability.

H.323

H.323 is an international multimedia conferencing protocol, developed by ITU-T (International Telecommunications Union) in 1996 for communication over Packet Switched Networks (LAN, WAN and Internet). H.323 is extensively used in VoIP (Voice over IP), Video Conferencing and Data Communication over the Internet. H.323 can manage failure of NEs (Network Entities) like Gatekeepers and endpoints. It also supports recovery of connection failures. H.323 performs coding in binary format that is appropriate for narrow and broad band connections.

SIP

SIP stands for Session Initiation Protocol developed by IETF (Internet Engineering Task Force) in 1999. It allows establishment of different sessions that can be used for communication over the Internet. SIP has no procedures defined to handle or manage failure of Network Entities.

In order to route a call effectively, certain issues should be considered and improved

accordingly such as:

Average Call Duration (ACD)

It is the total amount of time taken by the call. In case of lower ACD, it is expected that the quality of the connection is not good enough for the subscriber to continue the call. It is calculated as:

ACD = total call duration/ number of answered calls (i)

When the ACD is low, it is interpreted to mean that callers do not complete their calls as desired but rather hang up as a result of bad quality. For wholesale, this implies a loss in revenue minutes. In a retail service like Calling Cards, it is a very important issue since people would not like to purchase, for a second time, a calling card that gives a low ACD.

Post Dial Delay (PDD)

On dialing phone number, either there is a ring or busy tone that tells us that whether the called party is available or not. The time elapsed between dialing a number and hearing a tone is referred to as Post Dial Delay (PDD). In case of higher PDD, it is expected that there is no dial tone for the subscriber to initiate a call.

Answer-Seize Ratio (ASR)

It is the ratio between the successful calls and the attempted calls that cannot be answered for any reason. In case of lower ASR, it is expected that the route provided to the call is choked-up for the subscribers to make phone calls. It is calculated as:

ASR  = number of answered calls/number of call attempts (ii)

A low ASR implies that callers cannot get through to the other end, thus, signifying bad quality. Reasons for this are:

1. Congestion: There is more traffic than the available capacity. Therefore, some calls are blocked. 
2. Long Post-Dial Delay (PDD): PDD is the time between dialing a number and hearing the dial tone or busy signal. In general, when the PDD is greater than 10 sec. Callers consider it to mean that there is no connection and therefore hang up. This is seen as an unsuccessful attempt. 
3. Call Looping: This is the situation where a call moves from one carrier to another and back over and over again. In most cases, the call does not reach its destination. Even if it does, a long PDD is experienced. The call is interpreted as an attempt each time it comes back to the switch. Call looping usually involves two or more carriers and is very undesirable.
4. Dead air: Here, a call is being billed when no one is responding at the other end. This is very undesirable to the caller because he/she pays for a service he has not used.  
5. Call breaking: Caller and/or cal-lee get intermittent sound breaks. This is caused by impairments in the electronic equipment or packet loss in case of VoIP. 
6. Echo: This is the phenomenon whereby during a call, one repeatedly hears his/her own voice. In most cases, only one person hears an echo, which means his/her voice is reflected near to the other end-user. This occurs in VoIP.  
7. Fax failing: Fax which is data traffic, is transmitted using voice channels. It is described as failing when the receiver misses part or all of the transmitted data. 

Quality Measures

To establish a network that offers highest level of quality of service, telecom operators experience few challenges such as:

􀂾 Latency

It is the amount of time required to transmit data from source to the destination. It is an end-to-end delay that occurs in information exchange between two nodes. Simply, VoIP QoS Version 1.0 www.advancedvoip.com Page 5 of 8 it can be referred as the speed of the network that can affect the overall quality of the service. Delays in data packets can be reduced by reducing overall packet size.

􀂾 Jitter

Information is transferred from source to destinations via small messages called packets. Such packets experience certain delays to reach their required destinations. The variation in these delays is known as jitter and it adversely affect quality of the service provided. It makes certain sounds due to packet loss but can be managed via jitter buffers.

􀂾 Packet Loss

Data packets can be dropped due to congestion in the network or limited buffer size at the other end. Once these packets are lost, they cannot be recovered unless re-transmitted by the sender. Thus affect the speed and finally the quality of the network. To reduce data loss, QoS monitoring ensures congestion and queue management via various tools like Priority Queuing (PQ), Custom Queuing (CQ) etc. Queue management prevents queues from filling and provides space for high priority packets.

􀂾 Bandwidth

Bandwidth is the total capacity of a transmission medium to transfer data. Bandwidth and Latency both can affect the quality of service in terms of speed and capacity of the network. Greater the bandwidth more is the ability of the network to transmit data. Capacity of a network to transfer information decreases if the network is oversubscribed with users.