[Asterisk-Users] No sound with SIP Phones on the Internet

Alberto Forchino alberto.forchino at digitaltelevision.it
Wed Oct 15 05:21:54 MST 2003


Hi,
a little newbie question:
I've just installed asterisk and played a little with it. the server has a 
pubblic address while the clients (sjphone, msn messenger, sipset) are 
behind a firewall/NAT. sip part always works, while rtp part sometimes 
works, sometimes not. the question is: does asterisk decapsulates data 
coming from client 1 and re-encapsulates them in the call-leg from asterisk 
to client 2 so the problem could be caused by work overload in the server? 
or can it make data pass through?

the second question (that is related) is: I would like to send video and I 
set sip.conf with video. I saw that asterisk has plugins for h.261, h.263: 
what should I do if I want to send h.264 or MPEG4 for example? if asterisk 
decodes and re-encodes I MUST have a specific plugin, but if it only takes 
the data from the first call-leg to send them on the second one I don't 
need it...
thanks for any hint,
Alberto Forchino


On Mon, 13 Oct 2003 17:32:46 +0100, WipeOut <wipe_out at lycos.co.uk> wrote:

> Chris Hariga wrote:
>
>> This is bull... I can't believe that...
>> Must be a solution...
>>
>> Chris HARIGA
>>
>>
>> -----Original Message-----
>> From: asterisk-users-admin at lists.digium.com
>> [mailto:asterisk-users-admin at lists.digium.com]On Behalf Of WipeOut
>> Sent: Monday, October 13, 2003 9:57 AM
>> To: asterisk-users at lists.digium.com
>> Subject: Re: [Asterisk-Users] No sound with SIP Phones on the Internet
>>
>>
>> Chris Hariga wrote:
>>
>>
>>
>>> Yes, my Asterisk is behind a NAT but I forward all ports (100-56000) to 
>>> my Linux box.
>>>
>>>
>>>
>>>
>>>
>> There is your problem.. Asterisk does not like playing behind NAT.. The 
>> UA's can be made to work behind NAT but the server must have a public IP 
>> address..
>>
>>
>>
> There is a solution.. buy a SIP aware router with a built in SIP proxy.. 
> But even then you will probably still have issues..
>
> Search the archives and you will see that this issue has come up time and 
> time again and I have not heard of anyone who has managed to get Asterisk 
> to work correctly when the Asterisk server is behind NAT..
>
> If the SIP UA is also behind NAT then there is even less chance of it 
> working..
>
> Believe it, Don't believe it its your choice..
>
> Later..
>
>
>
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> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>



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