On 2014-05-28 6:39 AM, pae3 wrote:
Hi!
Don't miss RTP protocol :
pass proto tcp to port 20001
ah, no RTP will be moving over UDP.
--
Ted Bullock tbull...@northernartifex.com
On 2014-05-27, Andre Ruppert a...@in-telegence.net wrote:
You have two different protocols: SIP for signaling und RTP for media.
Media information between the endpoints is specified in SIP-SDP-packets
(session description protocol).
SDP-packets contain the original IPs of the VoIP-endpoints,
Hi!
Don't miss RTP protocol :
pass proto tcp to port 20001
Alex
On 05/27/2014 07:46 PM, Dmitry Petrakoff wrote:
Sorry, that was exactly I meant ( OT probably ):
The first issue with late hang-up most likely means, that calee hung up and his UAC sent
SIP BYE within existing
Does pf have specific rules for voip,
no
may be example of working pf_rule with voip?
I use a hardware phone (Linksys SPA 901),
a software SIP client (CSipSimple) on an Android,
and pjsua on OpenBSD, all behind OpenBSD NAT.
In pf.conf I let udp port sip and tcp port sip in, and anything
Does pf have specific rules for voip, may be example of working pf_rule
with voip?
Because for «standart rules» i have problems with voip.
set skip on lo
match out on pppoe0 from { em1:network } nat-to (pppoe0)
block
pass out
pass in on { em1 }
- after hanging up, the line near 3 minutes
On Tue, May 27, 2014 at 01:59:07PM +0400, Швецов Михаил wrote:
Does pf have specific rules for voip, may be example of working pf_rule with
voip?
Because for «standart rules» i have problems with voip.
set skip on lo
match out on pppoe0 from { em1:network } nat-to (pppoe0)
block
Hi!
It is most unlikely the issue of pf or its rules. Simply because your
issues are related to SIP (busy issue) and RTP/phone (voice volume).
Pf does not have any SIP ALG built-in so can't affect VoIP.
I'd like to suggest you to check busy issue with your VoIP provider or
to check out different
Am Dienstag, den 27.05.2014, 14:15 +0400 schrieb Dmitry Petrakoff:
It is most unlikely the issue of pf or its rules. Simply because your
issues are related to SIP (busy issue) and RTP/phone (voice volume).
Pf does not have any SIP ALG built-in so can't affect VoIP.
Well that is not completely
Tue, 27 May 2014 13:59:07 +0400
Швецов Михаил mv...@ya.ru wrote:
Does pf have specific rules for voip, may be example of working
pf_rule with voip?
Because for «standart rules» i have problems with voip.
set skip on lo
match out on pppoe0 from { em1:network } nat-to (pppoe0)
block
Sorry, that was exactly I meant ( OT probably ):
The first issue with late hang-up most likely means, that calee hung up and his
UAC sent SIP BYE within existing dialog. For some reasons either UAS on
caller's side or intermediate SIP proxy discarded that BYE. There could be the
same issue with
On Tue, May 27, 2014 at 01:59:07PM +0400, wrote:
Does pf have specific rules for voip, may be example of working
pf_rule with voip?
Because for «standart rules» i have problems with voip.
set skip on lo
match out on pppoe0 from { em1:network } nat-to (pppoe0)
On 2014/05/27 13:59, Швецов Михаил wrote:
Does pf have specific rules for voip, may be example of working pf_rule with
voip?
Because for «standart rules» i have problems with voip.
set skip on lo
match out on pppoe0 from { em1:network } nat-to (pppoe0)
block
pass out
pass in on
On Tue, May 27, 2014 at 3:33 PM, Stuart Henderson s...@spacehopper.org wrote:
It just workstm for me, no special setup needed, no static-port or anything,
just a standard nat-to rule. This is with various devices; snom and gigaset
hardware phones, softclient on android, pjsua on OpenBSD.
But
On Sun, Feb 25, 2007 at 08:45:25AM +0100, Julien TOUCHE wrote:
is there some plan to make pf voip-aware so it can handle dynamic port
allocation ?
i suppose answer risks to be no as else, there would be one for ftp (and
not ftp-proxy). So, what openbsd users use to add security to their voip
is there some plan to make pf voip-aware so it can handle dynamic port
allocation ?
i suppose answer risks to be no as else, there would be one for ftp (and
not ftp-proxy). So, what openbsd users use to add security to their voip
setup except pf and vlan ?
thanks
Regards
Julien
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