[asterisk-users] NAT/IPTABLES workarounds

[Sammy Govind]

Are you talking about having an SSH tunnel and route your SIP traffic
through it !!?

On Thu, Jan 5, 2012 at 4:20 AM, Kevin P. Fleming <kpfleming at digium.com>wrote:

> On 01/03/2012 10:03 AM, Patrick Lists wrote:
>
>> On 03-01-12 16:24, Danny Nicholas wrote:
>>
>>> Hello List,
>>>
>>> I work in an environment where I have to request IPTABLES changes rather
>>> than doing them myself. Is there a way to utilize my SSH (port 22) to
>>> get a functional (and with good sound) Asterisk installation with
>>> multiple channels up without requesting the 5060(SIP) 5061 (TLS) and
>>> UDP/RTP (usually 10001-20000) IPTABLES allowances?
>>>
>>
>> Not with SIP as it needs a port for signaling (usually 5060) and RTP
>> ports for sending the actual voice packets. So for SIP you will always
>> need multiple ports. If you can use IAX then you could use port 22 as
>> IAX only needs one port. The question is how are you going to SSH into
>> the box if you use the SSH port for Asterisk?
>>
>
> It is not practical (although not impossible) to run UDP over an SSH
> tunnel. Since VoIP media is generally transported over UDP, this will be a
> major obstacle.
>
> --
> Kevin P. Fleming
> Digium, Inc. | Director of Software Technologies
> Jabber: kfleming at digium.com | SIP: kpfleming at digium.com | Skype: kpfleming
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at www.digium.com & www.asterisk.org
>
>
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