On Mar 16, 2011, at 09:16, Rocco Radisch IT-Doc24 Ltd wrote:

> 
> 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 ?
> 

 I agree that Asterisk is popular  precisely for the reasons you mention. Any 
project that wants to do P2P calls (i.e. Skype Libre of sorts) needs to do two 
things IMHO: 
 - Do not require user to install server on their LAN (may be do this by 
growing community of server hosters)
 - Easy-to-use (SIP) client. 

 So yes Asterisk is too complicated for #1, but then the GNU falls over here as 
well. And methinks there is a gap in the market on that one...
 For #2, there is no shortage of SIP clients out there. 

> 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
> 

 Because we all love variety. But look closely at that list, and you will see 
that in terms of advanced call management (which clearly PBX users want), 
Asterisk leaves most of them in the dust. 

> 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
>>>>> 
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>>>> _______________________________________________
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>>> _______________________________________________
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>> _______________________________________________
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>> 
>> Send messages to this mailing list by addressing e-mails to: [email protected]
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>> 
>> 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 
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>> any way.
> _______________________________________________
> The Uganda Linux User Group: http://linux.or.ug
> 
> Send messages to this mailing list by addressing e-mails to: [email protected]
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> 
> 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.
> 

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