[asterisk-biz] Is ISP Blocking VoIP
trixter aka Bret McDanel
trixter at 0xdecafbad.com
Tue Jan 31 18:13:15 MST 2006
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) is a good resource for this.
--
Trixter 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
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: application/pgp-signature
Size: 189 bytes
Desc: This is a digitally signed message part
Url : http://lists.digium.com/pipermail/asterisk-biz/attachments/20060131/ae06255a/attachment.pgp
More information about the asterisk-biz
mailing list