Yenya's World

Fri, 03 Mar 2006

IP telephony

Playing with the IP telephony continues. I have reported few bugs in Ekiga, and it seems it makes big progress almost daily. I use the latest CVS version (from the ekiga snaphots site, wich also includes the snapshots of the libraries).

After few problems I now use Siproxd on my home NAT gateway, and I am able to call from the outside network to the laptop behind this NAT and vice versa. I have also obtained an account on the CESNET H.323 gateway, which gave me a public phone number for incoming calls. Unfortunately, the H.323 NAT/conntrack helper is still in the early stages of the development and not very stable, so I cannot use it from behind the NAT host. I hope I manage to fill a bug report soon.

I have also discovered few problems with CESNET VoIP - my H.323 number is not reachable from the Vodafone network, and some Cisco gear at CESNET does not like that my SIP From: header contains a non-numeric local part of the address (my login name instead of the phone number). The CESNET people are trying to fix both of these problems, and I must say, they were extremely helpful when I was trying to set up the VoIP service.

Yesterday a friend of my wife also tried a SIP telephone (Twinkle), and we were able to call her through the Freevoice.CZ gateway. Twinkle has some interesting features (such as two virtual voice lines), but it does not support H.323, and also the SPEEX codec (the later is in their TODO, though).

I wonder for how long the current pricing of the legacy voice services can remain at the current level, because the price of the broadband Net connection is about the same as the price of the analog voice line (especially in bigger cities on networks like Netbox), and the bandwidth is an order of magnitude (or two) higher. The monthly fee is about the same, but the legacy phone service is in addition to that charged for every minute of the call, while the Net connection is always on. The analog phone uses 64kbit/s for a single call, while SPEEX wideband codec needs about 32kbit/s, so the 1 Mbit/s line can hold 32 calls (and I am not counting the fact that for traffic inside the Netbox own network I have 10/10 Mbit/s of bandwidth). And as far as I can tell, the quality of the IP calls is not so bad, especially with the SPEEX codec.

So, is it time for huge price cuts, or time for yet another monopoly abuse like disabling the VoIP traffic on the telco's own network?

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