As we have seen in the previous blog article , there are major problems in carrying faxes over IP. Let’s see why Faxing over IP is a reliable standard for transmission of faxes.
A standard which makes the communication much more reliable, whenever conditions are not optional, is T.38.
The T.38 fax relay standard was devised in 1998 as a way to permit the transmission of faxes across IP networks between existing Group 3 (G3) fax terminals. T.38 carries T.30, the protocol used by faxes, over a packet-oriented connection.
Let’s examine which available solutions allow for the transmission of faxes over IP and the best practises that reduce related problems.
Fax over VoIP is a usual requirement in many deployments. Although the number of fax sessions per year is constantly decreasing, for the next ten years or so, we still need to include fax support in any Unified Communication platform.
Faxing was never meant to be used over the Internet, as it was designed specifically for transmissions over telephone lines with circuit connections and low latency.
A Unified Communication solution can integrate a software server fax. This software simulates a real fax machine, and allows you to send or receive an image file to / from a remote fax endpoint.
Fax Over RTP
Desired video codecs mainly fall between VP8 (and its evolution VP9) and H.264 and H.265. Other formats, such as H261 – H263 – H263p, can be mainly found in legacy conferencing systems and should be avoided.
Let’s examine the main differences between these codecs.
H.264 / H.265
H.264, also known as MPEG-4 Part 10, Advanced Video Coding (MPEG-4 AVC), is a block-oriented, motion-compensation-based video compression standard.
Wideband audio provides high-definition voice quality for telephony calls, and offers better quality than standard digital telephony. It does so by extending the frequency range of audio signals transmitted over telephone lines, resulting in higher quality speech.
The great advantages of wideband audio, also marketed as high-definition (HD) audio, are:
- Offers clearer sound quality
- Provides easier voice recognition, even in the presence of background noise, speakerphone, and voice overlapping (two persons speaking at the same time)
- Makes certain words easier to understand, such as words containing letters which sound similar
The final result for users is better phone calls, where both the effort required to understand each other and the need for callers to frequently repeat themselves is greatly reduced.
This time we will analyze what are the main characteristics to take into consideration when choosing codecs for communication systems.
Codec stands for coder-decoder. A codec codes a signal into a digital data stream and decodes a digital stream into a signal.
In the case of VoIP, when choosing a codec, we are mainly interested in two characteristics:
- Amount of bandwidth used
Adequate bandwidth is necessary for a high-quality conversation. Even when implementing jitter buffers and packet loss concealment in the endpoint, if the current connectivity cannot withstand the traffic transmitted over it, there is going to be huge packet loss, making it impossible for users to have a conversation.