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 From:  Daniele Guazzoni <daniele dot guazzoni at gcomm dot ch>
 To:  m0n0wall at lists dot m0n0 dot ch
 Subject:  Re: [m0n0wall] Voip Quality
 Date:  Thu, 18 Jan 2007 02:05:24 +0100
Paddy

the standard SIP port is UDP 5060 (can be also used on non-standard ports).
SIP is used for the parameter negotiation and call setup.
The negotiated parameter are as ex. codec-type, bitrate, dtmf-handling and most important the
communication channel.
As soon as both partners has agreed on a common set of parameters an RTP session is initiated.
RTP uses UDP as transport over dynamic agreed ports. Usually you can set a port range to use, to
ease to firewall setup.
As "quasi" standard the RTP range is 15000 to 20000. 
You can make it much smaller (at least one port per simultaneous call) if you own both ends,
otherwise check with your VoIP provider which range he implemented.

I you want to compare it to ISDN, SIP is the D-Channel (signalisation) and RTP are the B-Channels
(bearer) of a BRI.

Take a look at http://freshmeat.net/articles/view/2079/ it explains pretty well how SIP and RTP
works and what are the limitations when used over firewalls or with NAT.
Another good site for VoIP is http://www.voip-info.org

padraig at premierbb dot ie wrote:
> 
> I must admit, I am lost now. Am I right in saying sip ports are the 
> ports I said were 5060-5062? Are these only used start a voip call, and 
> then all voip traffic is carried on using different ports? What ports 
> should I specify? Or should I specify any. BTW I can only use a static 
> ip address as all this is routed through a wireless router.
> 
> Paddy
> ----- Original Message ----- From: "Daniele Guazzoni" 
> <daniele dot guazzoni at gcomm dot ch>
> To: <m0n0wall at lists dot m0n0 dot ch>
> Sent: Wednesday, January 17, 2007 1:42 AM
> Subject: Re: [m0n0wall] Voip Quality
> 
> 
>> Paddy
>>
>> you're talking about SIP ports but those are just signaling.
>> The actual VoIP communication is done on RTP and the ports are 
>> dynamically negotiated.
>> On most devices and PBX you can set a range of ports to be used for 
>> RTP (typically UDP 15000 - 20000).
>> You have to configure the traffic shaper for inbound / outbound SIP 
>> and RTP.
>> Otherwise you can establish the communication but then you could have 
>> a very worst quality.
>>
>> I use this setup with Asterisk (both SIP and IAX2) on an ADSL link and 
>> it works very well.
>>
>> PS: IAX is the better choice for WAN connectivity (NAT). It also 
>> provide a more efficient transport and bandwidth utilization.
>>
>>
>> Daniele
>>
>>
>> padraig at premierbb dot ie wrote:
>>> Hi
>>>
>>>
>>> I have posted this question on several bulletin boards, but all 
>>> suggestions failed to fix my problem. Basically I have a dsl 
>>> connection, then monowall, then the lan network consisting of 60+ 
>>> machines. What I am trying to do is prioritise port 5060-5062 for 
>>> voip traffic only. When the network is congested, I want the voip 
>>> quality to be as near as perfect as possible. could someone please 
>>> show me the "true" way of setting this up? BTW the dsl connection is 
>>> 6 meg down, 512k up.
>>>
>>> Paddy
>>
>>
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> 
> 

-- 


regards


-------------------------------------------------------------
Daniele Guazzoni
Senior Network Engineer, CCNP, CCNA


Linux and AMD-x86_64 or do you still with Windows and Intel ?

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