Quality of routes

Routes manage the path of a customer's outbound calls, while DIDs bring inbound traffic to the customer. They both take care of the transit of a SIP audio call from caller to callee, and have many of the same challenges to their quality in common.

White, black, and grey

The technical barrier for providing termination services (routes to PSTN) and origination services (DIDs that get calls from PSTN) is so low that in countries and regions where VoIP is under monopoly, or where a cartel of big companies control the market imposing hefty prices, the business opportunity is so compelling that a plethora of independent operators, of widely differing reliability and regulation compliance (or which are outright illegal) discreetly populate the scene.

Talking about routes and DIDs to and from these destinations, it is often referred to by the term "grey" market. That's because one side (you, the end customer) is white in the open, regulation abiding, while the other end is black in the dark, possibly illegal, side of the business. And of course you have all the 50 shades in between.

A white route or DID will go to a first tier operator or to the monopolist, and will have a robust price tag, but its audio quality, continuity, reliability and stability will be mostly assured. A service you can count on.

On the opposite side, the various shades of grey will be offered to you with costs that reflect quality and reliability, and some of them can stop working completely and forever without warning.

Many ITSPs organize their offers using grey routes where quality is not of paramount importance, backed up from second tier and first tier routes in case of cheap route failures.

Some ITSPs let customers choose a customized mix of routes of different qualities to different destinations, and a custom procedure to react to failures (for example, try another cheap route, escalate to second tier, and so on)

Codecs and bandwidth

Each voice call using G711 codecs (that is, native non-compressed telecom formats) consumes around 80-100 Kbps for each direction, including network overhead (for a possible total of around 200 Kbps). The G729 codec results in roughly 30 Kbps usage per direction, and is currently the most adopted VoIP codec because of its favorable balance between payload compression (low bandwidth usage) and audio perceived quality.

Bandwidth utilization can vary greatly depending on various factors such as SIP header compressions, network fragmentation, silence suppression, period of sample, and other minor details.

The sample duration at which the audio is packetized at is mostly 20 milliseconds, and, while this value is the most adopted because of its robustness in the face of packet loss and delays, the ratio between overhead (headers contained in the packet) and payload (actual encoded audio) is very unfavorable.

So, in situations where bandwidth is costly (for example, satellite communication, developing countries, radio transmission, and so on) the sample duration is often brought to 30, 40, or 60 msec, and/or other much more compressed codecs are used (for example, G723, ilbc, Speex), sacrificing some audio quality.

The quality of a voice call is determined by the worst quality of its path (for example, an ilbc originated call cannot get better because it's translated to your receiving G711; on the contrary, each translation further degrades the audio quality of the call).

For the same reason, check thoroughly, your ability to actually enjoy the advantages of a High Definition Audio Codec. If your call transits even for a moment on the PSTN, any HD will be reduced to worse than G711 (because of translation degradation). So high definition audio is mainly for calls to other SIP users. But even if you use an HD codec and your ITSP accepts it, it will not necessarily (actually almost never), be accepted as it is on the path from your ITSP to another ITSP, even if that second ITSP claims to support the same one. You can have better luck calling other customers of your same ITSP.

A special case of high definition audio implementation would be if your ITSP has a direct connection with 4G and LTE cellular network carriers. Many first tier cellphone carriers are beginning to roll out high definition audio to their customers, so ask your ITSP if it supports HD audio calls with them. Cellular networks, while they have been known since the beginning for way lower audio quality relative to PSTN, are about to invert this proportion and be the showcase for mass HD audio adoption.

Infrastructure capability

A very important issue with ITSPs is their propensity to overbook their bandwidth, or even worse, their capability to manage SIP signaling, for example, call establishment.

You can encounter a situation where your ITSP is growing so fast that it is not able to deliver enough bandwidth to all of its customers, particularly during peak time.

But more often the problem arises from the sudden arrival of a specific new customer, for example, a new call center, that moves all of its traffic into routes and servers that until now provided much less throughput.

Worst of all is when a customer using predictive dialers or teleblasters comes online. The bandwidth usage can be compatible with the ITSP setup, but the call attempted per second (cps) can bring the ITSP's SIP servers to a crawl, because for each successful cold call they try to connect to 20 numbers in old and ineffective lists. This can hamper your ability to place and receive calls.

Another, sometimes overlooked, limitation (on the customer side, this time) that can damage call quality and completion rate is the "Asymmetrical" world of ADSL. Asymmetrical Data Subscriber Line's bandwidth is, um, asymmetrical, while VoIP is completely symmetrical. So an ADSL can be pretty fast in downloading a video at 2 Mbps, but its VoIP bandwidth (and the number of maximum concurrent audio channels) will be defined by the upload speed, which is often dramatically lower than the download speed.

Packet delay and, worse, jitter (a discontinuous variation in delay that cannot be easily compensated) can negatively affect audio quality and is relative to the physical distance between end points and to the propagation delay in the path between them.

So, you can be better served by an ITSP, which SIP servers are connected to your SIP devices through a high speed or dedicated network (for example, MPLS), whose infrastructure is directly connected to a first tier Internet backbone, and which has its own datacenters near your region and near the region of your highest traffic.