3 ms·

Theoretically, yes. However, in practice, no. One of the great wins of packetization is that your infrastructure becomes generic in the sense that it is applica

Theoretically, yes. However, in practice, no. One of the great wins of packetization is that your infrastructure becomes generic in the sense that it is application-independent, allowing all subsequent improvements in eg. compression protocols to be applied retroactively. Therefore, if you would apply for example a variable bit rate, modern CODEC to the voice streams it's likely you would fit substantially more than a naïve set of voice channels on the raw PRI. The main costs are not in overhead, but enhanced session setup and round-trip latency and increased processing overheads at the edges. However, modern electronics offer embedded solutions making this ~free and a few ms are tolerable. IIRC voice is fine at <200ms latency, more or less, though less is obviously better. The philosophical battle was centralist vs decentralised, for-purpose vs generic infrastructure, single-mode vs. multi-mode network applications, circuit-switched guaranteed QoS vs. best-effort packets, waterfall vs. iterative development and investment. Packets won for general purpose networking, because the nature of physics and bureaucracy meant the cost and time savings for operators and users were substantial.

ssl-3 1y ago
Using a different codec can improve call-carrying capacity of any compatible link, for sure. But that's wiggling a different variable: Packetization-vs-not is a whole different game than g.711u vs g.723 is. And I agree, completely: There's very good reasons for packet-switched networks having won over TDM technologies like PRI, with IP to tie it altogether. What we've arrived at is beautifully generic in that the packets themselves don't care at all about what combination of lower layers were used to get them from A to B. However it gets there, it's just IP. That's really neat. Latency-wise, I'm a little torn: With the latency involved in a normal phone call on my normal cell phone today, I find that I "talk over" people on phone calls more than I did decades ago in the TDM days. The pacing is very different than it was. Sometimes, when the echo cancellation fails at some level, I can hear myself echo from the far end of a call. That echo is annoying, but I mention it because it is something that lets me hear the latency of the call. It's often on the order of 500ms RTT...which is quite a lot. We never had latency like that in the TDM days for domestic calls. (Historically, in TDM world: The voice data always arrived at the right time and there was also always a place to put it at the right time. It was real-time instead of best-effort, so there just wasn't any utility to having any large buffers along the way: The timing was either resolutely correct or it didn't work at all. Things are a lot mushier today.)

ssl-3 1y ago

Using a different codec can improve call-carrying capacity of any compatible link, for sure. But that's wiggling a different variable: Packetization-vs-not is a whole different game than g.711u vs g.723 is. And I agree, completely: There's very good reasons for packet-switched networks having won over TDM technologies like PRI, with IP to tie it altogether. What we've arrived at is beautifully generic in that the packets themselves don't care at all about what combination of lower layers were used to get them from A to B. However it gets there, it's just IP. That's really neat. Latency-wise, I'm a little torn: With the latency involved in a normal phone call on my normal cell phone today, I find that I "talk over" people on phone calls more than I did decades ago in the TDM days. The pacing is very different than it was. Sometimes, when the echo cancellation fails at some level, I can hear myself echo from the far end of a call. That echo is annoying, but I mention it because it is something that lets me hear the latency of the call. It's often on the order of 500ms RTT...which is quite a lot. We never had latency like that in the TDM days for domestic calls. (Historically, in TDM world: The voice data always arrived at the right time and there was also always a place to put it at the right time. It was real-time instead of best-effort, so there just wasn't any utility to having any large buffers along the way: The timing was either resolutely correct or it didn't work at all. Things are a lot mushier today.)