Is the delay on long distance interviews on TV real?

Question

Often when anchors in the studio are talking to reporters or people from a long distance, there is a small delay in the response, maybe one or two seconds.

This got me thinking, why is there such a delay?

I've observed the delay seems mostly independent of distance, from as little as half a second to as much as five seconds. Also, the transmission delays don't add up: you could send a broadcast around the Earth in about 500ms by pinging e.g. Japan from where I am, over the internet, and a video link is far simpler (with fixed latencies mostly.)

Perhaps it is a deliberate delay created by the reporter, so they have time to think about the answer to a question; or perhaps it is added by the studio so that in case a man-in-the-street interview goes a bit bad they can cut it off?

Answer 1

When broadcasting across a distance data needs to be compressed and it is buffered for this purpose. The buffer is a certain size, say 5 seconds at a time. When broadcasting live this buffering can cause overhead in the processing of the signal and can cause the signal to be delayed.

The codec needs a buffer because it analyzes video content over a period of time. For example, it analyzes movement or change from one video frame to the next in order to reduce the bit rate of the content as much as possible and produce high-quality images. To compress data at one point, it might use the analysis of data that occurs two seconds later. Therefore, the more data it can analyze, the better it can compress the stream. Reducing the buffer shortens the delay, but can result in lower quality images.

From Reducing Broadcast Delay.

In most cases this is unnoticeable due to modern technology. However when a reporter is on location they don't always have access to equipment to be able to stream without delay.

In some cases there is a deliberate delay included:

In radio and television, broadcast delay refers to the practice of intentionally delaying broadcast of live material. A short delay is often used to prevent profanity, bloopers, violence, or other undesirable material from making it to air, including more mundane problems such as technical malfunctions or coughing. In this instance, it is often referred to as a seven-second delay or profanity delay.

The best reference in the above article is this one.

These days the broadcasts can be seamless although when reporters are coming out of places like Libya etc. and they are using Skype or other internet video technology you will notice there is still a lag.

Answer 2

Some transmissions are routed via satellite: so there's a bigger end-to-end delay, due to the longer distance, due to targeting satellites which are in distant, geosynchronous orbits (instead of going the shortest route).

To get to geosynchronous orbit and back is necessarily a round trip of 100,000 km, or about 0.3 seconds (due to the speed of light).

I experienced such a round-trip delay myself, when I phoned Australia from England in the 1980s: before transcontinental, undersea cables (which route signals over much shorter distances than sending them into orbit and back) became more common.

Also, general-purpose IP networks (i.e. simple packet switched versus circuit switched networks) adds extra delay to real-time signals: because they're 'bursty', it's not enough for the signal to have enough bandwidth 'on average' ... you want the signal to get through all the time, and so they'll be using a jitter buffer somewhere, which adds an extra delay.

That's not to say that there isn't also some extra seconds for editorial/censorship purposes. But the best case for a conversation that's routed entirely via geosynchronous satellite is a (noticeable) delay of more than half a second (4 legs: the outgoing question beams up and then down, and then the interviewee's reply also beams up and then down).

Answer 3

Oftentimes the interviews are done over satellite uplinks. Normally, with geosynchronous satellites, so there is a bit of a delay due to the fact that the speed of light is annoyingly slow.

Answer 4

I worked on VoIP telephones systems, so apologies for limited references - this is from my working knowledge.

There are several sources of delay, and the human ear can start detecting it at very small values. (I saw a reference about 7-8 years ago that 20-30 milliseconds delay was enough to be noticeably annoying, although the delays on tools like Skype and mobile phones is typically much greater. Other people seem to have far more tolerance for this than me!) You will probably find that the delays that you find so annoying are generally only several hundred milliseconds - enough to disrupt conversation, as turn-taking cues are missing.

Sources for the delay include:

• Frame size: When your voice is sent over the Internet or your work network, it is broken up into "frames" of varying length, depending on the codec. 10ms is typical (e.g. G.729) but for lower-bandwidth applications frames may be 30ms. (e.g. G.723.1) The first part of the frame is not sent until the last part of the frame is recorded, so there is up to 30ms delay already. Your old-fashioned land-line didn't have this problem - it would be transmitting immediately.

• Speed-Of-Light Delay: As you point out, it can take a couple of hundred milliseconds for the signal to travel half-way around the world - more as it doesn't follow the shortest (Great Circle) distance. Others have pointed out that going via a geo-stationary satellite adds a huge additional delay - one that is normally avoided if possible. (I was told once that, back when satellites were more commonly used that, wherever possible, the satellite would only be used for one direction, and under-sea cables for the other, to reduce round-trip delays.)

• Congestion: If packets are sent over an internet connection there may be points at which they contend for bandwidth with other packets, adding delay. Even if there are no other packets to content with, each router along the way adds some processing delay.

• Jitter Buffer: Packets sent over the Internet are not guaranteed to all arrive in a set time. Some may be a few milliseconds faster or slower than others. In extreme cases, they can even be overtaken. This variance in delay is known as jitter. Any jitter in the actual playing of the sound causes the human ear to have trouble making out the words. So the receiver has to queue up the packets for a short period, to line them up evenly for smooth playing. (The jitter buffer turns jitter into delay, which is a compromise solution.) This jitter buffer size varies on the quality of the connection, but can easily add 60 ms to the delay.

• Algorithmic Delay: It takes time to process the compression - it is very CPU intensive, especially with transcoding, where it is necessary to convert from one compression format to another). Also, some compression techniques require "lookahead", which for G.723 adds an additional 7.5 ms delay. (Ref)

• If you are using a general-purpose operating system (e.g. a Windows laptop) that isn't made for real-time audio systems, you can expect to add some additional delays in the OS's and sound-card's responsiveness; these aren't the typical operations the OS is optimised to handle. They are tuned to run some longer jobs faster, not to handle very small jobs responsively.

Remember, when you see a person hesitating to respond, you are seeing the round-trip delay - therefore you need to double the total delays mentioned here.

Note: Any delays for beeping out obscenities and the like will occur after the conversation is recorded. That is, you will hear the broadcast a few seconds later, but they won't interfere with the conversation being held between the interviewer and interviewee. Adding a 5-7 second delay at that point would make the conversation untenable.