Hi guys, and girls, Male and Female,
do anybody know if anybody has made a implimentation of 3325 for ser?
or, is anybody makeing some form of support for this?
-Atle
Hi, I am trilling to make SER workin conectec to internet via a
router, the router only forwards the mensages to SER, the messages who
go to the 5060 port.
With this config i get SER working well locally, but when i try to
register from the outside i can't.
For more info I'm using sipsak for testing, and the version of ser is 0.9.3.
Last thing: P L E A S E H E L P
hello all,
not sure if everyone here is familiar with the madison river case:
http://informationweek.smallbizpipeline.com/60405214<http://www.tmcnet.com/tmcnet/articles/2005/voip-call-blocking-fcc-madison-r…>
not exactly the same thing, because they were blocking instead of charging
(they are a small rural CLEC who, like almost all phone companies, charge
less than cost for access and make their profit on minutes, and voip takes
away their minutes), but the principle is the same, and the FCC nailed them.
However, meanwhile the FCC chairman has changed, and who knows if the
general way of thinking is still the same...
at very least, it's good precendent.
as someone who works for a telecom regulator (i wont tell you which one and
where) i'd be very interested in what happens around the world. are there
countries where ISP's are not allowed to filter or otherwise interfere in
traffic?
I know there are companies that sell traffic filtering systems to ISP's so
they can prioritize and/or slow the performance of certain customers,
certain apps (filesharing and the like), etc. it's relatively trivial to
configure such systems to introduce massive packet loss to anything on port
5060, for example, or deprioritize UDP packets, or things of that nature.
-yair
I believe It is unlawful act and there must be some regulation, I am not
talking here that it will become known very well and this provider will
lost some customers... and some other provider will claim that they voip
friendly and get some more customer on it :-)
________________________________
From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On
Behalf Of Cesc
Sent: Wednesday, November 02, 2005 1:32 PM
To: Nils Ohlmeier
Cc: serusers(a)lists.iptel.org; Roger Lewau; serusers-bounces(a)lists.iptel.org; Iqbal
Subject: Re: [Serusers] voip charging by ISP's
Well, i think the ISP may care if they are also providing similar
bussiness. They won't probably be able to charge neither the voip user
or voip provider, but they can do some harm. Identifying the rtp streams
and sip signalling is not sooo difficult ... then, block a few messages,
add delay and jitter to the rtp (or udp in general ...) and you've got a
bad service, aparently coming from the voip provider.
There is software to detect and do all the above (one was presented in
VON this fall) ...
There was an article in IEEE Spectrum (sept, october, nov ... can't
remember).
And all the tricks Nils mentions ... hey, maybe the advanced user, but
my mom, no way! And then you need software that supports all those
tricks ... that is even further away. In this sense, i think carriers
have the winning side ...
Regards,
Cesc
On 11/2/05, Nils Ohlmeier <lists(a)ohlmeier.org> wrote:
On Wednesday 02 November 2005 14:28, Iqbal wrote:
> I agree, I mean the next step would be to charge ebay and yahoo per
> transaction, just wondering how easy it would be for ISP to setup a
> monitor and a billing model on this
I think the ISP should not care what it is transporting. The snail mail
also
does not look into what they are devlivering to charge you differently
(except express delivery but thats another story).
As soon as an ISP starts to look into the traffic from the user, the
users
will invent uncounted ways of fooling the ISP (use non-standard ports;
use
standard ports for the "wrong" service; use proxies; encrypt the
traffic; and
finally combine all this). All I can say to the ISP: happy debugging!
And in the end I would be interested in how a court would judge about
the
outcoming bills (who has to prove what?) :-)
Nils
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Hi all,
is it possible to manipulate the From-Header (similiar to strip() and
prefix()), before relaying a INVITE message? If so, how do I do this?
Regards,
Stefan
Hi
i was looking into a model where ISP may charge (SBC is already thiking
of charging Vonage :-)), now if an ISP wanted to charge, all they would
need is to extract the INVITE, BYE and all the stuff in the middle, to
do some billing per VoIP provider sending traffic on there network, is
anyone aware of any sniffers which could do this not from a proxy, but
from the network level at a ISP end.
Iqbal
Hi all,
i just committed specs for debian to generate both tls and non-tls
packages - thanks to Klaus for support ;).
I also generated the packages for debian unstable - see :
http://openser.org/pub/openser/latest/packages/deb-unstable/
in the next days I hope packages for different versions of debian will
be uploaded.
regards,
bogdan
Valued Friends,
I am trying to process 5xx response codes and to resend the INVITE to an
alternative VoIP gateway upon receiving an 5xx response from a VoIP
gateway. So, I need to somehow figure out the IP address of the VoIP
gateway from which SER receives the 5xx response.
I tested and I receive 503 response codes only in onreply_route
and not in failure_route at all.
The 503 response does not include any "contact" fields that Would
indicate the IP address of the VoIP gateway from which The 503 response
is being received at SER.
Do you know how I can figure out the IP address of the VoIP gateway from
which SER is receiving the 503 response?!
Thanks
ramin
hi
can anyone explain me why i get the error mentioned in below trace. Please let me know how can i resolve this.
SIP/2.0 477 Unfortunately error on sending to next hop occurred (477/TM)
Call-ID: 21b297bc85242e507848f6c297cf4c3f(a)69.229.23.54
CSeq: 1 REGISTER
From: "test11" <sip:test11@iptel.org:5060;transport=tcp>;tag=6626965
To: "test11" <sip:test11@iptel.org:5060;transport=tcp>;tag=321967ba5deb3e7f484a2df055818800-a111
Via: SIP/2.0/TCP 69.229.23.54:5060;branch=z9hG4bK67a6e74fa0f95b090b70ff5d3d969e73;rport=6719
Server: Sip EXpress router (0.9.0udpfifo (i386/linux))
Warning: 392 195.37.77.99:5060 "Noisy feedback tells: pid=17793 req_src_ip=69.229.23.54 req_src_port=6719 in_uri=sip:69.229.23.57:5060;transport=tcp out_uri=sip:69.229.23.57:5060;transport=tcp via_cnt==1"
Content-Length: 0
Thanks
Rajesh
---------------------------------
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I am looking for the best way to remove headers from INVITES sent to my
PSTN gateway. The invites from my UA are too long and return '513's when
they include authentication. My provider doesn't support BYOD so getting
them to change maxlength is not an option. there are several headers
that are sent by my ua that are not sent by the ua they have provided.
when I strip the extra headers out using textops subst function it
leaves an empty line which the pstn gateway doesn't seem to like very
much. Is there some other method to strip headers? or maybe my regular
expression syntax is wrong? the re I use to select the Date: header for
removal is '^Date:.*$' replaced with // . I am testing my re's using
sed. I must admit I know nearly nothing about using regular expressions.
Thanks,
Dale