To add on the list,
- Skype can adjust and change codecs during a call without disruptions.
That depends on bandwidth load, congestion level, cpu load etc which is
quite clever. Same with video codecs. Not that I looked for it
extensively but I am not aware of an open-source based solution which
can do the same at the moment. I think here lies the huge success of
Skype, to be able to cope with any environment and the constant changes
during a call.
- Next question would be about conferencing capabilities on audio and
video if GNU SIP Witch only connects endpoints. SIP can stream but needs
to be implemented on client side I guess.
Paul,
as I understood GNU SIP Witch doesn't handle audio (media stream) at
all. It initiates connection between the two endpoints and deals with P2P.
"GNU SIP Witch is a destination router for the SIP protocol. This means
it is primarily concern is not in making things interconnect “with” the
SIP Witch Server, like say something like Asterisk does very well, but
rather instead is designed to enable two (or more) endpoints to find and
then directly connect with each other."
(Asterisk can do something similar via canreinvite or directrtpsetup).
"And as of 1.8.0 it does ZRTP and TLS"
GNU SIP Witch has ZRTP support as well:
"This project’s definition of secure media is similar to Zimmermann’s
work on ZRTP, in that we assure there is no forwarding knowledge by
using uniquely generated keys for each communication session. "
On the Asterisk / P2P issue, I came across an interesting tool
yesterday. Haven't had the time to read further and check for
suitability of SIP and test it.
The tool makes two side NAT traversal "magically" possible. Connecting
two nodes directly without going through a 3th party proxy (redirection).
http://samy.pl/pwnat/
Rocco
On 15/03/2011 1:32 PM, Paul Bagyenda wrote:
This was announced 14th March it seems. Today, I receive a Skype update
notification, to take me from v2.0 to v5.0. Coincidence? May be.
I see some problems with GNU SIP Witch:
- Doesn't do audio transcoding. This is one of those places where Asterisk
shines. One side might be speaking u-law, the other side GSM, and they'll work
together nicely.
- Not really P2P, though it claims to be. This is where it'd beat Asterisk.
At least in terms of setup.
Asterisk works fine. The code is [expletive deleted] but it works. And as of
1.8.0 it does ZRTP and TLS (for SIP signalling traffic) so you can do secure
calls. (See for example privatewave.com who have a mobile SIP client that does
secure calls over 3G data using just this feature.)
P.
On Mar 15, 2011, at 12:55, Martin Atukunda wrote:
Hi Lug,
FYI, New project "GNU Free Call" Announced. This could be a skype competitor.
Here is an excerpt from the site / blog entry:
-- 8< --
Our goal is to make GNU Free Call ubiquitous in a manner and level of usability
similar to Skype, that is, usable on all platforms, and directly by the general
public for all manner of secure communication between known and anonymous
parties, but without requiring a central service provider to register with,
without using insecure source secret binary protocols that may have back-doors,
and without having network control points of any kind that can be exploited or
abused by external parties. By doing so as a self organizing meshed calling
network, we further eliminate potential service control points such as through
explicit routing peers even if networks are isolated in civil emergencies.
-- >8 --
See http://planet.gnu.org/gnutelephony/?p=14
- Martin -
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