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kimotho
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SIP One Way Audio & How to enable ports on NAT Firewall

Fri Jul 27, 2007 2:12 pm

Hi,

Am using rb532 indoor board with v2.9.42 OS working fine except Sip to Sip calls are not working on Linksys PAP2 v1 firmware 3.1.22. When i call PAPB from PAPA, i cant hear the PAPB but PAPB can hear me but faintly.

I've enable NAT mapping & NAT support on the Linksys and now i can make SIP 2 SIP calls fine but the sound quality is terrible. I suspect the NAT Firewall on the RouterOS is affecting this. How can i disable the firewall or better still enable this ports on routerOS? Is the beta v3 OS better in handling SIP calls?

SIP signalling ports (UDP) = 5060 - 5061
DNS port (UDP) = 53
TFTP port (UDP) = 69
RTP/RTCP ports (UDP) = 10000 - 30000

Thanx in advance
-Kim
 
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sergejs
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Re: SIP One Way Audio & How to enable ports on NAT Firewall

Mon Jul 30, 2007 4:04 pm

3.0 version provides support for SIP helper over NAT environment.
Currently you need to use one-to-one NAT for this,
http://wiki.mikrotik.com/wiki/How_to_li ... Local_ones
You can get information about SIP and NAT issues here,
http://www.voip-info.org/wiki-Asterisk+ ... +solutions
 
andycrazy
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Re: SIP One Way Audio & How to enable ports on NAT Firewall

Mon Mar 03, 2008 8:06 pm

my english is very poor
ip firewall service-port pr
Flags: X - disabled, I - invalid
# NAME PORTS
0 ftp 21
1 tftp 69
2 irc 6667
3 h323
4 sip 5060 5061
with this rule work fine in mikrotik 3.3 put in service ports sip 5060 and 5061
 
changeip
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Re: SIP One Way Audio & How to enable ports on NAT Firewall

Mon Mar 03, 2008 8:34 pm

3.0 version provides support for SIP helper over NAT environment.
Currently you need to use one-to-one NAT for this,
Wait a minute ... one to one nat? You typically dont even need a SIP helper if you have 1-to-1 nat because all ports are forwarded that way. Really ? So what's the SIP helper doing then, just changing the private IP for the public one in the udp/5060 payload ? I thought the SIP helper was supposed to enable connection tracking for the RTP/RTSP which runs on other ports as well, similiar to the FTP helper.
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Letni
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Re: SIP One Way Audio & How to enable ports on NAT Firewall

Tue Mar 04, 2008 4:01 am

I was having a one way audio issue in V3 NAT until I disabled the SIP helper... Go figure.

-Louis
 
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airstream
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Re: SIP One Way Audio & How to enable ports on NAT Firewall

Sun May 25, 2008 1:55 am

I was having a one way audio issue in V3 NAT until I disabled the SIP helper... Go figure.

-Louis
Actually we have had such fun getting our SIP to work after upgrading to V3.9

Thanks for the tip to disable SIP and H323 in the "Service Ports" of firewall. Now it all works again (seems to), thats a go figure from me too.
Keep tryin' its bound to work!!
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Re: SIP One Way Audio & How to enable ports on NAT Firewall

Fri May 30, 2008 12:12 am

This is very interesting. I would like to know more information about this. We just tried this and same thing, it worked.
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kthameen
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Re: SIP One Way Audio & How to enable ports on NAT Firewall

Fri May 30, 2008 12:24 am

Same Problem when SIP was disabled on v3.9 Both SIP and Media were working well.
 
wpeople
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Re: SIP One Way Audio & How to enable ports on NAT Firewall

Wed Jul 09, 2008 5:57 pm

I'm using asterisk server (192.168.58.x) behind Mikrotik NAT (192.168.1.1). (asterisk has local voip clients on internal - another ethernet - interface, on 192.168.1.0/24)

udp/5060 and udp/10000-20000 forwarded to asterisk

Before disabling the firewall/services/sip:
---------------------------------------
Remote clients (coming from internet thru mikrotik) can talk to local clients, asterisk, and call call out, using landline telephony interface.
Voip provider can be used by local phones.
Remote clients cannot use voip provider (no audio -RTP- at all, in any direction!)

After disabling the firewall/services/sip:
-------------------------------------
Remote clients can do anything as before, plus can use Voip provider!

ps: when using ROS 3.6 or 3.8 (this board originally come with any of these) Asterisk cannot even registery to voip provider (until upgrade to 3.10). (i've not tried to disable SIP service)

Hope this helped to anyone

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