what are the options for very low bandwidth PC-to-PC VoIP?
September 11, 2008 1:13 AM Subscribe
I'm looking for a software PC-to-PC VoIP program that will handle an extremely low bandwidth connection with high UDP packet loss. This is to reach a person at the far end of a highly contended (8:1 or greater) TDMA VSAT connection. Total bandwidth available to the client PC at the far end of the satellite link is about 64 kbps. Skype does not work properly due to >45% UDP packet loss and high latency jitter.
We need something that moves about 3.5KB/s or less of data each direction while providing a moderately decent voice call quality. It does not need to be able to reach the PSTN or accept incoming calls from phones, strictly PC-to-PC functionality. Nothing like SkypeOut is required. Can anyone suggest software that will do this? I know that Yahoo Messenger and the latest Windows Live Messenger (formerly MSN messenger) do video and audio-only chats, but I'm not sure about the minimum bandwidth requirements.
The alternative is of course text-only chat (Pidgin, Adium etc connected to various networks) or IRC, which works flawlessly, but we'd really like to be able to establish a voice connection.
We need something that moves about 3.5KB/s or less of data each direction while providing a moderately decent voice call quality. It does not need to be able to reach the PSTN or accept incoming calls from phones, strictly PC-to-PC functionality. Nothing like SkypeOut is required. Can anyone suggest software that will do this? I know that Yahoo Messenger and the latest Windows Live Messenger (formerly MSN messenger) do video and audio-only chats, but I'm not sure about the minimum bandwidth requirements.
The alternative is of course text-only chat (Pidgin, Adium etc connected to various networks) or IRC, which works flawlessly, but we'd really like to be able to establish a voice connection.
Specifically, I see a 3.4 kbit Speex option which sounds like what you'd need.
posted by Rhomboid at 2:32 AM on September 11, 2008
posted by Rhomboid at 2:32 AM on September 11, 2008
you don't have much choice. something that supports speex is probably going to be the best IF you can enforce the frame size. lpc10 would work if you don't mind sounding like a robot.
also keep in mind that a satellite link has hell of latency, so your conversations will have lots of lag time.
none of the above is really going to allow decent voice quality or overall call quality.
posted by dorian at 8:17 AM on September 11, 2008
also keep in mind that a satellite link has hell of latency, so your conversations will have lots of lag time.
none of the above is really going to allow decent voice quality or overall call quality.
posted by dorian at 8:17 AM on September 11, 2008
To be brutally honest, you're out of luck - I very much doubt any software will make an acceptable VOIP connection over a link with 45% packet loss. VOIP codecs (including low-bitrate ones like Speex and iLBC) are generally designed to cope with some packet loss, either by just dropping a moment of audio or "concealing" it by extending the last received sound. The data that was in those lost packets is never resent or otherwise recovered. That means that 45% of the sound in your conversation won't make it across the link, and nothing the codecs can do will fix that. You won't be able to carry on any kind of intelligible conversation.
Beyond the packet loss, the latency is going to be brutal - even a couple of hundred ms is enough to make conversation seriously difficult, and you'll be well over that on a satellite connection. It might be possible to write custom software that would improve the packet loss situation by sending each packet multiple times or something, but the latency would still kill you.
(Sorry if you already knew this, but packet loss isn't specific to a protocol like UDP - it affects all protocols equally. The reason you don't notice it on text-only chat connections is that they use TCP, which automatically detects packet loss and retransmits lost data.)
posted by pocams at 11:09 AM on September 11, 2008
Beyond the packet loss, the latency is going to be brutal - even a couple of hundred ms is enough to make conversation seriously difficult, and you'll be well over that on a satellite connection. It might be possible to write custom software that would improve the packet loss situation by sending each packet multiple times or something, but the latency would still kill you.
(Sorry if you already knew this, but packet loss isn't specific to a protocol like UDP - it affects all protocols equally. The reason you don't notice it on text-only chat connections is that they use TCP, which automatically detects packet loss and retransmits lost data.)
posted by pocams at 11:09 AM on September 11, 2008
oh lord yes, the packet loss is going to kill it. even if you found something that used SIPS (secure and runs over TCP) the encryption would increase the bitrate (by a smallish amount but in your case enough to make a big difference probably) and the TCP error correction would increase the lag beyond the already likely unacceptable levels.
or you could try to run the calls inside e.g. OpenVPN but that's also going to add overhead and lag.
you're almost better at this point, recording very compressed/lossy mp3 and emailing them to each other. no, seriously.
posted by dorian at 3:00 PM on September 11, 2008
or you could try to run the calls inside e.g. OpenVPN but that's also going to add overhead and lag.
you're almost better at this point, recording very compressed/lossy mp3 and emailing them to each other. no, seriously.
posted by dorian at 3:00 PM on September 11, 2008
This thread is closed to new comments.
posted by Rhomboid at 2:28 AM on September 11, 2008