https://www.opensips-solutions.com
>https://www.siphub.com
>
> On 23.09.2024 17:36, Robert Dyck wrote:
> > I thought perhaps nat_traversal had been abandoned. In nathelper the flags
> > were changed to strings but not so in nat_traversal.
> >
> > On Sunday, Sept
u have a 100% registrations driven platform, it will
> not make too much of a difference for you.
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>https://www.opensips-solutions.com
>https://www.siphub.com
>
> On 18.09.2024
I am reaching out to users and developers.
I read that nat_traversal was supposed to replace nathelper. It appears that
they have co-existed for many versions now. Nat_traversal is supposed to
overcome NAT issues in a multi proxy environment.
Looking at the functions provided by the two modules
the diff comes for the actual
> logging. What are the 2 versions you tested ?
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>https://www.opensips-solutions.com
>https://www.siphub.com
>
> On 10.04.2024 00:21, Robert Dyck wrote:
> > In the past
enSIPS Founder and Developer
>https://www.opensips-solutions.com
>https://www.siphub.com
>
> On 10.04.2024 00:21, Robert Dyck wrote:
> > In the past I would insert xlog with $mb into my script for debugging
> > purposes. Now I find that the messag
In the past I would insert xlog with $mb into my script for debugging purposes.
Now I find
that the message buffer output is not being formatted.
Instead of
Message Buffer
REGISTER sip:192.168.1.2 SIP/2.0
Via: SIP/2.0/UDP
192.168.1.4:5070;branch=z9hG4bKce021e1d6a292d1504d0ff89e60c9ba;rport
I
I was too hasty. The subscriber has a long expiry on it's subscribe ( 1 hr )
and it is not configurable. The entry was eventually deleted.
Currently using opensips-3.4.3. I am experimenting with using resource lists
for presence. I have a question about table rls_presentity. When a UA
subscribe
Currently using opensips-3.4.3. I am experimenting with using resource lists
for presence. I have a question about table rls_presentity. When a UA
subscribes to a presentity using an entry in a resource list an entry is
created in the DB table rls_presentity. When the presentity sends publish it
While running opensips in debug mode I noticed that for initial requests of
dialog creating
methods the debug logs were showing a To tag where none actually exists. The
tag
displayed was actually the From tag.
INVITE sip:8@192.168.1.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK.k
olutions.com <https://www.opensips-solutions.com/>
> > https://www.siphub.com <https://www.siphub.com/>
> >
> > On 11/20/23 11:11 PM, Adrian Georgescu wrote:
> >> XCAP is a failure. Not that we did not try, it was a bad idea and it
> >> failed.
>
The context here is subscription to presence by way of a resource list. The
learning curve is steep. I have read the tutorial. The tutorial gives an
example of a
rls-service xml document. In the example the resource list is contained within
the
services document. Various other examples I have
I forgot to mention that nat_uac_test should use at least flag 2. This insures
that usrloc contains "Received".
On Thursday, January 19, 2023 10:27:47 A.M. PST nutxase via Users wrote:
> Hi guys
>
> So i notice when i register a WSS client to opensips the contact shows
> something like Contact":
Do you have opensips-cli setup? If so, do "mi ul_dump". Look for your
"invalid" contact.
Your should have a line like "Received": "sip:1.2.3.4:60310;transport=wss",
The IP address should be routable.
On Thursday, January 19, 2023 10:27:47 A.M. PST nutxase via Users wrote:
> Hi guys
>
> So i noti
This is a Linphone problem. A simple UA has no way of knowing that all the
other UAs are unable to take the call. The Linphone developers should be
encouraged to change the default response to a rejected call. Perhaps a
universal reject could be an option but not the default.
On Tuesday, Octobe
p to
> $T_branch_idx. You can do the same thing for replies, and that should
> cover all cases.
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-solutions.com
>
> On 1/27/22 19:23, Robert Dyck wrote:
> > Opensips adds its via ( w
I am interested in trying the rtp_relay module but the documentation about the
$rtp_relay pseudo-variable seems sparse. This variable can become quite
complex with several components some of which have sub-components. In
particular the flags, peer and delete components could have several parts.
Opensips adds its via ( with branch info ) after script processing but before
forwarding. Opensips branch info is not available to the script when
processing an INVITE. I have attached some text of an INVITE with rtpengine
and with "offer via-branch=1". What rtpengine receives is the branch para
sistent "per branch", so you can rely
> on that.
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>https://www.opensips-solutions.com
> OpenSIPS eBootcamp 2021
>https://opensips.org/training/OpenSIPS_eBootcamp_2021/
>
> O
t;per branch", so you can rely
> on that.
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>https://www.opensips-solutions.com
> OpenSIPS eBootcamp 2021
>https://opensips.org/training/OpenSIPS_eBootcamp_2021/
>
> On 1/6/2
t; (after forking) is unique and consistent "per branch", so you can rely
> on that.
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>https://www.opensips-solutions.com
> OpenSIPS eBootcamp 2021
>https://opensips.org/training/Open
Attached here is a prettier version of the three offers.>From opensips
Jan 1 10:03:57 [2670144] Invite with first via host 192.168.1.2 and branch ID
z9hG4bKd83e.3a8b6577.0
Jan 1 10:03:57 [2670144] WebRTC-legacy interworking
Jan 1 10:03:57 [2670144] The answer profile must be opposite of the of
I am reaching out to the users out there to help me figure out why I get
occasional call failures when it involves rtpengine and forked calls. Calls
involving rtpengine but not forked are solid. For instance there is no problem
with a call between a SIPified WEBRTC phone and some end of life dev
s://www.opensips-solutions.com
> OpenSIPS eBootcamp 2021
>https://opensips.org/training/OpenSIPS_eBootcamp_2021/
>
> On 11/11/21 10:34 PM, Robert Dyck wrote:
> > The module documentation for msilo gives us an example of
> > configuration to deploy the service.
> >
ing/OpenSIPS_eBootcamp_2021/
>
> On 11/13/21 12:12 AM, Robert Dyck wrote:
> > How does one set the time stamp that openips prefixes to an offline
> > message that is sent when the UA registers?
> >
> > 2021-11-12 14:06 from 5
https://opensips.org/training/OpenSIPS_eBootcamp_2021/
>
> On 11/16/21 5:54 PM, Robert Dyck wrote:
> > I think I saw a report of a seg fault.
> >
> > On Tuesday, November 16, 2021 7:52:26 A.M. PST Bogdan-Andrei Iancu wrote:
> >> What kind of difficulties with 3
gt; OpenSIPS eBootcamp 2021
>https://opensips.org/training/OpenSIPS_eBootcamp_2021/
>
> On 11/16/21 5:42 PM, Robert Dyck wrote:
> > Still on 3.2.2. I saw reports of difficulty with 3.2.3.
> > Should I be recompiling?
> > Thanks, Rob
> >
> > On Tuesday, Nov
aining/OpenSIPS_eBootcamp_2021/
>
> On 11/13/21 12:12 AM, Robert Dyck wrote:
> > How does one set the time stamp that openips prefixes to an offline
> > message that is sent when the UA registers?
> >
> > 2021-11-12 14:06 from 5
How does one set the time stamp that openips prefixes to an offline message
that
is sent when the UA registers?
2021-11-12 14:06 from 5 "[Offline message - Wed Dec 31 16:00:00 1969 ] HI THERE"
Thanks, Rob
___
Users mailing list
Users@lists.opensips
The module documentation for msilo gives us an example of configuration to
deploy the
service.
In a block staring with "if(!lookup("location"))" we see the following --
# if the downstream UA does not support MESSAGE requests
# go to failure_route[1]
t_on_if (!db_does_uri_e
Opensips doesn't care about media. However rtpengine can bridge DTLS to non-
DTLS.
On Friday, December 25, 2020 12:30:22 P.M. PST Ali Alawi wrote:
> Hello,
>
> Is there a way to use DTLS on opensips through openssl?
>
> Any help or guid would be appreciated.
net/tcp_common.c will not compile. Too many errors to list
here.
It seems to be a new file as compared to my 3.0.2.
Rob
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Context opensips-3.0.2
The TCP protocol enables support for ping/pong by default. It
is the underlying protocol for websocket. I am seeing short
messages from webrtc UAs at 10 second intervals. Opensips
is rejecting the messages.
Sep 4 11:15:30 [3091728] DBG:proto_wss:ws_process: Using
the g
Unfortunately I was hasty in my interpretation. There is definitely a problem
but it lies
with the UA and not opensips. The UA mis-identifies itself and the caller sends
the
ACK to the wrong UA.
On Saturday, August 15, 2020 8:39:20 A.M. PDT Robert Dyck wrote:
I should explain the
My database was created with the old opensipsdbctl tool. The database
engine is sqlite. I want to start using opensips-cli to administer the
subscriber table. The trouble I am having is opensips-cli wants to connect to
the database named "opensips". How do I associate the sqlite database at /
us
I should explain the consequence of this error.
A and B register with the same AOR. A receives the correct
instance ID while B receives A's ID. There is a call to the AOR. B
answers the call and sends 200 OK and identifies itself
incorrectly. Caller receives 200 OK and sends an ACK to the
insta
Two UAs with the same AOR register. Both support GRUU. Both are assigned the
same sip
instance..
0fb66f5c-90f4-4611-9141-2594480977aa
SIP/2.0 200 OK
received=2001::9B5D;rport=46004;branch=z9hG4bK598182 To: ;tag=59de.372db74950592a93c1f2e8ff9432ad9f From: "test" ;tag=q3obuoma57 Call-ID: cphnnph
Robert Dyck wrote:
Perhaps you misinterpreted my wording. I actually tired SDES=off but crypto
attributes were still
inserted..
It is a bit strange that ICE=force should arbitrarily add the crypto.
On Monday, July 6, 2020 12:25:44 A.M. PDT Răzvan Crainea wrote:> You should use
the SDES-
ine.
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-solutions.com
>
> On 7/4/20 10:34 PM, Robert Dyck wrote:
> > I have run into an issue with rtpengine and the ICE=force option.
> >
> > To quote the rtpengine README
&g
I have run into an issue with rtpengine and the ICE=force option.
To quote the rtpengine README
With `force`, ICE attributes are first stripped, then new attributes are
When using the force option where I think it will be appropriate I found it
also adds crypto
attributes. I believe this inv
using the
source address. Now that the binding request is going to the UA I have to find
where the binding
response is going.
On Friday, July 3, 2020 8:06:16 P.M. PDT Robert Dyck wrote:
While configuring my script for rtpengine I got a rather confusing result. The
test involved a UA
tethered
While configuring my script for rtpengine I got a rather confusing result. The
test involved
a UA tethered to a phone with only IPV4 availble. The test was a call to a UA
registered
with an IPV6 address. The call was answered successfully but there was no
media. A
sniffer at the rtpengine hos
the "received" field. Call routing will fail. This applies whether
or not the contact is
nated when GRUU is in use. Perhaps uncommon once but always present in WEBRTC.
On Thursday, June 25, 2020 9:56:39 A.M. PDT Robert Dyck wrote:
I have submitted bug #2154.
I believe it is a re
I have submitted bug #2154.
I believe it is a registration problem. The "received" should never be null in
the location table.
On Wednesday, June 24, 2020 2:36:04 P.M. PDT Robert Dyck wrote:
Context: opensips 3.0.2
I wanted to cleanup a working configuration so I eliminated the NA
Context: opensips 3.0.2
I wanted to cleanup a working configuration so I eliminated the NAT check if
the address family
was IPV6. This was in the initial request route. I was surprised to see that an
IPV6 INVITE would
fail. The REGISTERs were good.
Could someone explain to me what is happenin
ok at the README file.
Based on the flags, rtpengine can bridge encrypted RTP traffic to unencrypted
RTP traffic. It can
also do transcoding.
So yes, it plays man-in-the-middle :)
Regards,
Ovidiu Sas
On Tue, May 19, 2020 at 18:32 Robert Dyck wrote:
Perhaps someone with knowledge of the
, May 16, 2020 at 3:37 PM Robert Dyck wrote:>> I
am wanting
to convert my config/script to use rtpengine instead of rtpproxy.> I think it
would better deal
with webrtc. After looking at some examples I> found, I see a couple of
parameters that are not
mentioned in the opensips>
I am wanting to convert my config/script to use rtpengine instead of rtpproxy.
I think it would better deal with webrtc. After looking at some examples I
found, I see a couple of parameters that are not mentioned in the opensips
documentation. First there is the offer/answer option ice=force-rel
Regarding opensips-3.0
Use case - webrtc client behind NAT
The rtpproxy module emitted the error message "can't extract media port from
the message" ( by the way, very misleading ). In reality extract_mediainfo
fails
because it could not find a supported payload type in the media description.
The following configuration snippet worked for me in 2.4 but causes a coredump
in 3.0.1 and
3.0.2.
In request route ( sequential )
xlog("Check for GRUU, Method is $rm\n");
Results
Mar 06 13:39:37 slim.mylan /usr/local/sbin/opensips[62100]: *Check for GRUU,
Method is BYE*
*Req
With opensips 3.0 the new tool for accessing opensips is opensips-cli. The
database module of opensips-cli only accepts the SQL variants. Does this mean
that dbtext will in the future be deprecated? Eventually not supported?
Rob
___
Users mailing lis
Never mind. A stupid mistake on my part. I should have just copied from my
working 2.4.5
instead of relying on faulty memory.
On Monday, October 14, 2019 7:32:18 P.M. PDT Robert Dyck wrote:
I am test driving 3.0.1. Using menuconfig I compiled the core, the default
modules and extra
modules
I am test driving 3.0.1. Using menuconfig I compiled the core, the default
modules and extra
modules mysql, presence, presence_xml and xcap.
Using menuconfig I generated a residential script with auth and presence. I
want to use db_text
with this minimal installation. I tweaked the configurati
Using the ip transform to resolve the address worked for me. When I get around
to it I should do it on opensips start up and cache the address
On Thursday, July 18, 2019 7:53:41 A.M. PDT Vitalii Aleksandrov wrote:
> Hi,
>
> Original question was about different thing but the destination IP of a
Thank you Yuri Ritvin
{ip.resolve} transform works for me. The example given in the documentation is
misleading. You can't use a literal string. You need to put into a var of some
sort and then transform it.
On Thursday, June 27, 2019 3:35:37 P.M. PDT rob.d...@telus.net wrote:
> On second thou
h_db.html#func_www_authorize[1]
Bogdan-Andrei IancuOpenSIPS Founder and Developer
http://www.opensips-solutions.com[2]OpenSIPS Bootcamp 2018
http://opensips.org/training/
OpenSIPS_Bootcamp_2018/[3]
On 11/14/2018 08:03 PM, Robert Dyck wrote:
I added "
I added "modparam("auth_db", "use_domain", 1)" but it doesn't make a difference
to the
subscriber table.
On Wednesday, November 14, 2018 9:36:34 AM PST Robert Dyck wrote:
[root@slim opensips]# opensipsctl add abc xyz
*new user
ensips-solutions.com[1]OpenSIPS Bootcamp 2018
http://opensips.org/training/
OpenSIPS_Bootcamp_2018/[2]
On 11/14/2018 06:52 PM, Robert Dyck wrote:
I do not have that parameter set and I do not use multiple domains.
The problem was that after I cor
http://opensips.org/training/
OpenSIPS_Bootcamp_2018/[3]
On 11/07/2018 10:09 PM, Robert Dyck wrote:
My understanding is that GRUU processing in opensips is automatic, provided it
is not disabled.
No further configuration or scripting is required. Is that co
/html/docs/modules/2.4.x/auth_db.html#param_use_domain[1]
Bogdan-Andrei IancuOpenSIPS Founder and Developer
http://www.opensips-solutions.com[2]OpenSIPS Bootcamp 2018
http://opensips.org/training/
OpenSIPS_Bootcamp_2018/[3]
On 11/07/2018 04:30 AM, Robert Dyck wrote:
Just a guess. Try
if $tu {remove("location","$tu");}
Not tested. A nonzero value may evaluate as TRUE.
On Tuesday, November 13, 2018 12:56:42 AM PST Pasan Meemaduma via Users wrote:
Hey,
Anyone have a suggestion for this?
On Thursday, 8 November 2018, 8:09:50 AM GMT+5:30, Pa
After some thought I realized that a lookup had to be invoked while in dialog.
The BYE was
directed at the proxy and the GRUU needed to be mapped to the device that was
the intended
target.
Added the following to script for "in dialog"
xlog("Check for GRUU, Method is $r
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
My understanding is that GRUU processing in opensips is automatic, provided it
is not disabled.
No further configuration or scripting is required. Is that correct.
A GRUU capable UA rergisters and receives public and temporary GR identities.
The UA
establishes a dialog with another UA. The ca
I have updated my small test bed from 2.3.2 to 2.4.2. I didn't bother to back
up the
'subscriber" table and it was wiped by the installation. No big deal, it was
tiny.
So I added the users but made an error.
opensipsctl add abc xyz -- I didn't specify the domain. The UAC would not
register.
ing credentials collected from network level.
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>http://www.opensips-solutions.com
> OpenSIPS Summit 2018
>http://www.opensips.org/events/Summit-2018Amsterdam
>
> On 01/11/2018 01:59 AM
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>http://www.opensips-solutions.com
> OpenSIPS Summit 2018
>http://www.opensips.org/events/Summit-2018Amsterdam
>
> On 01/09/2018 05:53 PM, Robert Dyck wrote:
> > Let me rephrase. The UA receives a 401 me
re-usage is checked.
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>http://www.opensips-solutions.com
> OpenSIPS Summit 2018
>http://www.opensips.org/events/Summit-2018Amsterdam
>
> On 01/08/2018 08:36 PM, Robert Dyck wrote:
> >
Using opensips 2.3.2 compiled from source
I have a buggy UA that insists on reusing a stale nonce. I tried to
work around it by setting disable_nonce_check. It didn't work for
me. Am I misunderstanding the purpose of the parameter or is this
an opensips bug?
Jan 8 09:46:19 [11380] DBG:core:se
I am reporting here because I don't know how to leave a bug report with Sippy
Software.
When I installed opensips and configured it use use rtpproxy and the unix
socket, opensips would not start. It turned out to be an issue with
permissions. It was simple to change to the loopback instead but
I had a working opensips 1.11 installed by Fedora package manager. I have been
trying opensips 2.2.2 and decided to make the jump. Since 2.2.2 is not
available as a package from Fedora I cloned the source.
I have tried installing using menuconfig and also make but neither installs the
necessary
SIPS Founder and Developer
> http://www.opensips-solutions.com
>
> On 07.12.2016 22:43, Robert Dyck wrote:
> > Actually when I was testing I entered the listen parameters on the command
> > line. I have now added them to the configuration file. The result is the
> &g
OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
> On 05.12.2016 21:37, Robert Dyck wrote:
> > This is a re-send. I apologize if it is a duplicate. I suspect my
> > subscription was not enabled.
> >
> >
This is a re-send. I apologize if it is a duplicate. I suspect my subscription
was not enabled.
--
I have been doing some testing with opensips 2.2.2 using ipv6. I found that
the server will only respond to a re
I have been doing some testing with opensips 2.2.2 using ipv6. I found that
the server will only respond to a request over IPV6 if it is configured to
listen on the default port.
Wireshark sees a request addressed to the server but there is no reply.
Running opensips in the foreground show no a
Răzvan Crainea wrote:
> Hi, Robert!
>
> See my answers inline.
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Solutions
> www.opensips-solutions.com
>
> On 11/08/2016 02:15 AM, Robert Dyck wrote:
> > I have some question regarding rtpproxy capabilities in
I have some question regarding rtpproxy capabilities in relation to IPV4-IPV6
interworking.
The articles I have read say that you need to assign an address from each
address family to rtpproxy. They go on to say that rtpproxy will then be in
bridged mode. Others define bridge mode as assigning
s,
> >
> > Bogdan-Andrei Iancu
> > OpenSIPS Founder and Developer
> > http://www.opensips-solutions.com
> >
> > On 20.10.2016 00:35, Robert Dyck wrote:
> >> Thanks for your input. The second scenario doesn't appear to be an
> >> issue.
> >>
gt; Regards,
> >
> > Bogdan-Andrei Iancu
> > OpenSIPS Founder and Developer
> > http://www.opensips-solutions.com
> >
> > On 20.10.2016 00:35, Robert Dyck wrote:
> >> Thanks for your input. The second scenario doesn't appear to be an
> >&g
restarting - OpenSIPS cannot change listeners during runtime.
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
> On 19.10.2016 19:14, Robert Dyck wrote:
> > Using 1.10.5
> > My ISP provides an IPV6 p
Using 1.10.5
My ISP provides an IPV6 prefix which unfortunately is not static. My address
does not change spontaneously but if the host is down for a significant time
the address will change.
I thought it would be simply solved by specifying a listening interface in the
configuration file. Unfo
Maybe before fiddling with the contact make sure you are not being hit by a
bug in some versions of asterisk. The re-invite arrived OK but not the ACK.
Was the route set present in the ACK. In-dialog messages should not alter the
route set. The UAS is not required to send a route header in the r
Is it masks in general that you do not understand? Using KCalc for example we
see that 2112 decimal is 1100 in binary and 2048 is 1000.
"Apply" usually means AND the two values. In this example the flag will match
one of the bits in the mask.
On Thursday 21 May 2009, Ricardo Mar
ll add on the TODO list the possibility to do a register at
> reply time, so that we can use opensips as mid-registrar.
>
> Regards,
> Bogdan
>
> Robert Dyck wrote:
> > I like the idea that we could maintain a local registrar that accurately
> > reflects the remote regist
I like the idea that we could maintain a local registrar that accurately
reflects the remote registration. I would go so far as to say that it would
be useful to be able to optimize opensips as an ALG. Some people could use a
proxy as an edge device on a LAN. You could have several phones with t
Very peculiar. The RURI was rewritten with the URI in the Route header. Does
your script rewrite it? This usually only happens when the route set shows a
next hop and the next hop is a strict router. If loose routing is in effect
all the way, the RURI does not change.
On Thursday 12 March 2009,
Additionally SDP can be sent in an ACK following 200 OK when the INVITE did
not include it.
On Sunday 08 March 2009, Alex Balashov wrote:
> It means you are applying the NAT UAC test function for SDP to a request
> that does not have an SDP payload.
>
> It should only be applied to messages that
set.
I am sorry about the confusion. You are quite right. The UA does not send a
R-R in a request.
On Thursday 05 March 2009, Iñaki Baz Castillo wrote:
> 2009/3/4 Robert Dyck :
> > Twinkle as one example however does not
> > send R-R with the in-dialog request.
>
> Hi, no o
As an aside, there are Asterisk machines out there that do not follow the
rules. Specifically they will alter their route set according to R-R received
in-dialog. This usually does not cause a problem because most UA's repeat
their R-R in the in-dialog request. Twinkle as one example however doe
Do you control the UAs at location A? Do they use STUN or are they configured
to masquerade as the router's public address? Both would claim to be
listening on the same IP. Many ( most? ) routers do not support hairpin
routing.
On Wednesday 17 December 2008, troxlinux wrote:
> Hi list, I want t
an and for whom they provide the
> benefit.
>
> Adrian
>
> On Dec 10, 2008, at 9:32 PM, Robert Dyck wrote:
> > I see a need for a very basic proxy-like B2BUA. This would
> > completely hide the
> > local topology. This would provide privacy and extra security as
at you want out of OpenSIPs. It's not a
> B2BUA and does not mask topology. I'd love to hear other's who don't
> believe this.
>
> www.opensipstack.org has an open source SBC, but I can't vouch for
> it's capabilities. You may also want to check out FreeSw
I see a need for a very basic proxy-like B2BUA. This would completely hide the
local topology. This would provide privacy and extra security as well as
working around the bad behaviour of some service providers.
Rob
On Wednesday 10 December 2008, Brett Nemeroff wrote:
> For what it's worth, I've
92 matches
Mail list logo
