[asterisk-biz] Is ISP Blocking VoIP

[Sergey Kuznetsov]

I see the only one way to do it.
To make simple test program which will be started on both endpoints.
Both boxes do the ntpdate right before the tests.
then one box sends and receives the number of rtp packets with 
increasing sequence numbers and stores the timestamps in the log
the same is done on the second box, it sends and receives from/to the 
first box and stores the sequence numbers and timestamps into the log.
then they exchanges by the logs, and third perl script compares the 
result and shows lost packets and latency for each packet in text or 
graphics.



All the Best!
Sergey.

Script Head wrote:
> Can anyone name a couple of reasonably priced or freeware tools that 
> could help me measure the performance between two VoIP endpoints?
>
> As far as ISPs introducing jitter to VoIP calls, can't this be solved 
> by passing VoIP over a VPN?
>
> ScriptHead
>
>
> On 1/31/06, *trixter aka Bret McDanel* <trixter at 0xdecafbad.com 
> <mailto:trixter at 0xdecafbad.com>> wrote:
>
>     On Tue, 2006-01-31 at 16:22 -0800, Rusty Shackleford wrote:
>     > I have every reason to believe that this is exactly what is
>     happening.
>     > Our company has some VOIP "extensions" from our IP PBX that live
>     at the
>     > end of Comcast cable modems. Within 3 minutes of commencing a
>     call, the
>     > audio quality (on the upstream leg) goes to hell (it actually sounds
>     > more like packet loss). This will continue for a variable period of
>
>     I want to reiterate that I have not read up on this so I may be
>     wrong on
>     the legal aspects, but its my understanding the FCC has not ruled on
>     jitter inducing gear, and that its technically legal because its not
>     blocking.  The blocking is afaik only limited to ISPs that offer the
>     service themselves - to say otherwise would put ISPs that filter adult
>     content at risk.  But again I havent read anything on either
>     aspect from
>     the FCC itself, so please dont rely on my for accuracy in this matter.
>
>     The hardware I have seen advertised is a 'black box' that ISPs can
>     install and it qill intentionally cause jitter on voice payloads
>     and it
>     came out just after when I heard the FCC ruled that ISPs cant block
>     VoIP.  Interesting coincidence isnt it?
>
>     As I recall the case the FCC ruled on (and again I did not hear it
>     from
>     an authoritative source but instead a friend) it was a large ISP
>     (I want
>     to say verizon but that may not be right) was blocking to force people
>     to use their VoIP alternative.  Adding jitter to your competition can
>     have a similar effect as only that ISPs service would appear good.
>
>     IIRC I posted a link to the device inquestion and a news story
>     about it
>     to either this list or to users 2-3 months ago, but I am on medication
>     right now so my memory may not be what it should.
>
>     The archives should yield more information, again google may be your
>     friend.
>
>     Sorry for being vague about all of this, but I am not in a position to
>     research this currently, I did however want to add bits of information
>     that might be useful to help others research it and possibly find an
>     answer.
>
>
>     There are FOSS (I believe FOSS anyway) tools to measure VoIP packet
>     performance.  I think there are tools that will even do it on non VoIP
>     stuff so you can compare the difference between RTP and say some
>     random
>     game that uses UDP (I think it might be unfair if the packet types
>     were
>     different as they may have different priorities for other reasons).
>     These tools may help analyze the problem and tell you if you have high
>     amounts of jitter or dropped packets on RTP but not on something else.
>     source forge (sf.net <http://sf.net>) is a good resource for this.
>
>
>     --
>     Trixter http://www.0xdecafbad.com <http://www.0xdecafbad.com>    
>     Bret McDanel
>     UK +44 870 340 4605   Germany +49 801 777 555 3402
>     US +1 360 207 0479 or +1 516 687 5200
>     FreeWorldDialup: 635378
>     http://www.sacaug.org/ Sacramento Asterisk Users Group
>
>
>     -----BEGIN PGP SIGNATURE-----
>     Version: GnuPG v1.4.2 (GNU/Linux)
>
>     iD8DBQBD4Asr+1olxlzQw5cRAtJuAJ9bkgaAf+wvAwxIFgJPuh3MykIlvwCfe8zH
>     T/wX3MHyEOIgYxEt/YndWsY=
>     =n/tX
>     -----END PGP SIGNATURE-----
>
>
>     _______________________________________________
>     --Bandwidth and Colocation provided by Easynews.com
>     <http://Easynews.com> --
>
>     asterisk-biz mailing list
>     To UNSUBSCRIBE or update options visit:
>        http://lists.digium.com/mailman/listinfo/asterisk-biz
>
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-biz mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-biz
>