Products

Voice connectivity can provide significant issues for IT specialists attempting to use SIP with smaller datapipes / congested networks or behind a NAT.

Traditionally these issues have been the Achilles heel of Internet voice communications.  It always seems one way audio or lost registrations issues plagued voice networks that were not able to assign public IP addressed to each phone.

At this point you spent hours or days trouble shooting and testing to make sure your voice communications work only to find yet another issue. This is why VoiceBus developed the VoicePath a small stand-alone network appliance that automatically surveys your network, and communicates complex details back to a central point which can then be analysed by our network engineers. The VoicePath will then provision itself automatically to met your network's unique needs, acting act as a RTP proxy, assisting with firewall rules, providing QoS, phone provisioning and much more.  Deploying your voice network has never been this easy!  Best of all the VoicePath is codec agnostic, so you can unleash the power of high definition voice.

To give you an example of how bandwidth limits will impact your communication, take a look at the follow details about codec with relation to bandwidth consumption.

Many DSL packages will offer an uplink of 768 kb/s, which comes out to be 98 KB/s.  Make sure to note the case of the k's.  Lower case k's are kilobits, while K's are kiloBytes.  Take a look at the chart below to see what one channel of a phone conversation will consume with respective codec.

Narrowband VoIP codecs
Codec Payload Bandwidth Description
G.711 64 KB/s This is the most universally supported codec used in IP telephony. Most phones will try to use this by default. This narrowband codec supports frequencies in the 300 to 3,400 hertz range and is uncompressed. Although the quality is very good, it consumes a lot of bandwidth.
G.729 8 KB/s This is the second most supported codec and offers nearly the same quality as G.711. The key advantage is that it is compressed eight times smaller than G.711 while sounding almost as good. This is licensed, so use of the codec is not free.

Wideband VoIP codecs
Codec Payload bandwidth Description
G.722 48 to 64 KB/s This is the most common wideband codec available in IP phones, though wideband support is only recently gaining momentum. The quality is excellent at twice the sampling rate of standard G.711, but the compression isn't that great. But considering the fact that it's the same bit rate as narrowband G.711 but delivers much more realistic sound, G.722 will be one of the best codecs to use if your IP telephones will support it. Wideband supports frequencies of 50 to 7,000 hertz. There are no longer any patents covering G.722, so it's free for anyone to use.
G.722.1 16 to 32 KB/s This is a wideband codec, aka Siren7, developed by Polycom. Its key advantage is that it's a computationally efficient and compact codec at 16 to 32 kbps, which is less than half the bandwidth required by G.722.  16 kbps mode isn't appropriate for noisy audio input or if music is mixed in, since the compression artifacts are noticeable.  32 kbps is good for any kind of workload.  This codec must be licensed from Polycom, and it's currently used only in Polycom's high-end video conferencing systems under the marketing name "Ultimate HD." Current Polycom IP phones use the marketing term "HD Voice," which supports only generic G.722 for its wideband codec, although future IP phone models may support G.722.1.
G.722.2 6.6 to 23.85 KB/s Also known as AMR-WB, this is a wideband codec. A 6.6 kbps mode is also supported, but 12.75 is the practical bit rate for speech in a clean environment. The higher 23.85 bit rate is better for noisy conditions and music. At the time of this writing, I'm not aware of any IP phones that support this codec. It's currently used by T-Mobile in Germany for cell phone applications.
Speex wideband 10 to 28 KB/s This is an excellent open source codec from Speex that offers very good wideband quality at relatively low bit rates. VBR (variable bit rate) is also supported in 12 or 18 kbps mode. This is a free open source codec and can be used by anyone. It's supported by open source Asterisk PBX, but no hard IP phones and only one soft IP phone support it.
Siren14 ultra-wideband 24 to 32 KB/s Siren14 is a free-to-license (not to be confused with license-free) ultra-wideband version of the G.722.1 codec from Polycom. Some of the high-end features in G.722.1, like echo cancellation and noise reduction, are omitted. Even though this is royalty free, you still need a license from Polycom. Siren14 supports a wider dynamic range up to 14,000 hertz, compared with 7,000 hertz in wideband.
Siren22 ultra-wideband 32 to 64 KB/s Siren22 is a proprietary ultra-wideband codec from Polycom that currently can't be licensed. Siren22 supports an even wider dynamic range--up to 22,000 hertz. The digital sampling rate is 48 KHz.