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vincent.delporte at bi...
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 Posted: Wed Jan 16, 2008 12:25 pm Post subject: [asterisk-users] [IAX] Up-to-date list of soft- and hardphon |
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Hello
There's a lot of information on VoIP at www.voip-info.org ...
but there's also a lot of outdated information there as well :-/
Since SIP is a pain to use when NAT is involved, especially when both
the Asterisk server and the remote phones are behind NAT... I'd like
to try IAX to see how it works and if it solves the issue.
I'd like to start with a softphone (Windows only), and then, if tests
prove successfully, buy a hardphone. What would be your
recommendations?
Thank you. |
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asterisk.org at sedwar...
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| Posted: Wed Jan 16, 2008 12:43 pm Post subject: [asterisk-users] [IAX] Up-to-date list of soft- and hardphon |
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On Wed, 16 Jan 2008, Vincent wrote:
| Quote: | Hello
There's a lot of information on VoIP at www.voip-info.org ...
but there's also a lot of outdated information there as well :-/
Since SIP is a pain to use when NAT is involved, especially when both
the Asterisk server and the remote phones are behind NAT... I'd like
to try IAX to see how it works and if it solves the issue.
I'd like to start with a softphone (Windows only), and then, if tests
prove successfully, buy a hardphone. What would be your
recommendations?
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Diax is probably the smallest Windows softphone.
I like my Digium Iaxy and a POTS instrument.
Thanks in advance,
------------------------------------------------------------------------
Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000 |
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gordon+asterisk at dro...
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| Posted: Wed Jan 16, 2008 1:08 pm Post subject: [asterisk-users] [IAX] Up-to-date list of soft- and hardphon |
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On Wed, 16 Jan 2008, Vincent wrote:
| Quote: | Hello
There's a lot of information on VoIP at www.voip-info.org ...
but there's also a lot of outdated information there as well :-/
Since SIP is a pain to use when NAT is involved, especially when both
the Asterisk server and the remote phones are behind NAT... I'd like
to try IAX to see how it works and if it solves the issue.
I'd like to start with a softphone (Windows only), and then, if tests
prove successfully, buy a hardphone. What would be your
recommendations?
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IDEFISK or Zoiper as it's called now.
However, you'll need to do similar things to your asterisk box & router if
it's behind NAT for IAX as you do for SIP. (You will need a static IP
address on the NAT router and port-forward 4569 to the asterisk box, just
as you'd port-forward 5060 and 10000-20000 for SIP)
And a SIP phone behind a NAT router is also solvable if it supports STUN.
I know that SIP behind NAT isn't perfect, but with care, it's very usable
and workable. I have many installations doing just this, as I'm sure many
others on the list have too.
Gordon |
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vincent.delporte at bi...
Guest
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| Posted: Wed Jan 16, 2008 5:01 pm Post subject: [asterisk-users] [IAX] Up-to-date list of soft- and hardphon |
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On Wed, 16 Jan 2008 18:08:23 +0000 (GMT), Gordon Henderson
<gordon+asterisk at drogon.net> wrote:
| Quote: | However, you'll need to do similar things to your asterisk box & router if
it's behind NAT for IAX as you do for SIP. (You will need a static IP
address on the NAT router and port-forward 4569 to the asterisk box, just
as you'd port-forward 5060 and 10000-20000 for SIP)
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Am I wrong to understand that IAX only needs one port, TCP4569 by
default? So I only need one port for each phone, while SIP requires at
least 3 (SIP, and one RTP each way)?
| Quote: | And a SIP phone behind a NAT router is also solvable if it supports STUN.
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But not all NAT routers support STUN, ie. keeping UDP ports open so
that incoming packets can make it.
| Quote: | I know that SIP behind NAT isn't perfect, but with care, it's very usable
and workable
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But unless I'm mistaken, when NAT is involved, canreinvite must be set
to no, ie. all RTP packets must go through Asterisk instead of flowing
from one phone to the other?
Thanks guys. |
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Claudio.Cavalera at it...
Guest
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| Posted: Thu Jan 17, 2008 6:20 am Post subject: [asterisk-users] [IAX] Up-to-date list of soft- and hardphon |
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| Quote: | -----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
Gordon Henderson
However, you'll need to do similar things to your asterisk
box & router if
it's behind NAT for IAX as you do for SIP. (You will need a static IP
address on the NAT router and port-forward 4569 to the
asterisk box, just
as you'd port-forward 5060 and 10000-20000 for SIP)
|
Please correct me if I'm wrong, for Iax clients you don't need to do
static port-forwarding as they will create upon registration one entry
in NAT table with UDP port for both signalling and media. On the other
hand, sip clients (without Stun) are difficult to manage behind Nat
because of RTP/RTCP ports.
I don't want to start a flame Iax vs Sip, just to clarify respective
advantages.
Best Regards,
Claudio
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dfullertasterisk at sh...
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| Posted: Thu Jan 17, 2008 1:09 pm Post subject: [asterisk-users] [IAX] Up-to-date list of soft- and hardphon |
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Cavalera Claudio Luigi wrote:
| Quote: | | Quote: | -----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
Gordon Henderson
However, you'll need to do similar things to your asterisk
box & router if
it's behind NAT for IAX as you do for SIP. (You will need a static IP
address on the NAT router and port-forward 4569 to the
asterisk box, just
as you'd port-forward 5060 and 10000-20000 for SIP)
|
Please correct me if I'm wrong, for Iax clients you don't need to do
static port-forwarding as they will create upon registration one entry
in NAT table with UDP port for both signalling and media. On the other
hand, sip clients (without Stun) are difficult to manage behind Nat
because of RTP/RTCP ports.
I don't want to start a flame Iax vs Sip, just to clarify respective
advantages.
Best Regards,
Claudio
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I believe you are correct, as long as the client sends *something* to
the server at frequent enough intervals that the router keeps the
connection in it's active list.
-Dave |
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timebandit001 at gmail...
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