Your standard SIP session does not typically mediate audio between the two
end-points. The server merely sets up the session. There is a way to get the
audio to pass through the server (e.g. for transcoding or NAT traversal
reasons), and Asterisk uses these.
The standard behaviour of Asterisk is to hold two audio (rtp) stream
during a call. One to each endpoint. With the canreinvite and
directrtpsetup options it is possible to connect both caller endpoints
directly after initiating the call via Asterisk. Both options have
various requirements to make this work though. In most cases (speaking
of Asterisk users) that is not desired. E.g. It wouldn't be possible to
record the call in case the audio stream doesn't pass through Asterisk
(only at the endpoint).
I think the major difference is that Asterisks main functionality lies
in the underlying call plan programming. Making it easy to replace a
conventional PBX and do hundreds of things on top which wouldn't be
possible with a hardware PBX.
But, going back to GNU Free Call, Asterisk simply for connecting end
points? Isn't that overkill? I think a SIP proxy server serves a better
purpose, such as OpenSER or GNU SIP Witch in this case.
What is your opinion on this ?
Since you are in b*tch mode :-) Any idea how many Asterisk alike
open-source software is out there? Why do we have all these if Asterisk
is the non-plus ultra ? ;-)
http://en.wikipedia.org/wiki/List_of_SIP_software
http://en.wikipedia.org/wiki/Comparison_of_VoIP_software
Cheers,
Rocco
On 15/03/2011 3:20 PM, Paul Bagyenda wrote:
Your standard SIP session does not typically mediate audio between the two
end-points. The server merely sets up the session. There is a way to get the
audio to pass through the server (e.g. for transcoding or NAT traversal
reasons), and Asterisk uses these.
For the ZRTP& TLS, yes, it is true GNU SIP Witch supports it, but then the
issue is so does Asterisk, so what's the point of a new project except may be to
amuse ourselves? (Plus, since I am in b*tch mode: They mention one of the deadest
projects I ever had the misfortune of hoping on all those years ago: Bayonne. Not a
good sign.)
But the first issue you mention has been one of my pet peeves for years now.
On the audio side this is actually kinda solved, if you use the AMR-NB codec
(for which there is an Asterisk 1.8 patch) which can vary the bit rate on the
go. For video, ahem. Only now is this being taken anywhere near seriously, and
then quite shabbily.
P.
On Mar 15, 2011, at 14:58, Rocco Radisch IT-Doc24 Ltd wrote:
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 -
_______________________________________________
The Uganda Linux User Group: http://linux.or.ug
Send messages to this mailing list by addressing e-mails to: [email protected]
Mailing list archives: http://www.mail-archive.com/[email protected]/
Mailing list settings: http://kym.net/mailman/listinfo/lug
To unsubscribe: http://kym.net/mailman/options/lug
The Uganda LUG mailing list is generously hosted by INFOCOM:
http://www.infocom.co.ug/
The above comments and data are owned by whoever posted them (including
attachments if any). The mailing list host is not responsible for them in any
way.
_______________________________________________
The Uganda Linux User Group: http://linux.or.ug
Send messages to this mailing list by addressing e-mails to: [email protected]
Mailing list archives: http://www.mail-archive.com/[email protected]/
Mailing list settings: http://kym.net/mailman/listinfo/lug
To unsubscribe: http://kym.net/mailman/options/lug
The Uganda LUG mailing list is generously hosted by INFOCOM:
http://www.infocom.co.ug/
The above comments and data are owned by whoever posted them (including
attachments if any). The mailing list host is not responsible for them in any
way.
_______________________________________________
The Uganda Linux User Group: http://linux.or.ug
Send messages to this mailing list by addressing e-mails to: [email protected]
Mailing list archives: http://www.mail-archive.com/[email protected]/
Mailing list settings: http://kym.net/mailman/listinfo/lug
To unsubscribe: http://kym.net/mailman/options/lug
The Uganda LUG mailing list is generously hosted by INFOCOM:
http://www.infocom.co.ug/
The above comments and data are owned by whoever posted them (including
attachments if any). The mailing list host is not responsible for them in any
way.
_______________________________________________
The Uganda Linux User Group: http://linux.or.ug
Send messages to this mailing list by addressing e-mails to: [email protected]
Mailing list archives: http://www.mail-archive.com/[email protected]/
Mailing list settings: http://kym.net/mailman/listinfo/lug
To unsubscribe: http://kym.net/mailman/options/lug
The Uganda LUG mailing list is generously hosted by INFOCOM:
http://www.infocom.co.ug/
The above comments and data are owned by whoever posted them (including
attachments if any). The mailing list host is not responsible for them in any
way.
_______________________________________________
The Uganda Linux User Group: http://linux.or.ug
Send messages to this mailing list by addressing e-mails to: [email protected]
Mailing list archives: http://www.mail-archive.com/[email protected]/
Mailing list settings: http://kym.net/mailman/listinfo/lug
To unsubscribe: http://kym.net/mailman/options/lug
The Uganda LUG mailing list is generously hosted by INFOCOM:
http://www.infocom.co.ug/
The above comments and data are owned by whoever posted them (including
attachments if any). The mailing list host is not responsible for them in any
way.
