[Freeswitch-users] SIP Multiplexed Trunk as IAX2 or TDM

Tue Jan 31 20:36:31 MSK 2012

I have pondered this topic for years and have never come to a
conclusion I was happy with to move on it.

Basically, on one hand, as Ken mentioned if you increase the ptime of
every call, you get a lot of the same benefits of trunking at the cost
of more audio lost if a packet disappears.  This is not a big risk
even at 60ms its only a minor loss of audio.  The trunking approach
does offer much less latency but its a matter of deciding if a complex
implementation of trunking is worth the few ms of latency.

If you look at trunking as a whole, its the idea of muxing in as many
calls bound to the same destination into one stream to avoid overhead.
 One problem on the internet is that many devices very tightly obey a
small MTU and even drop packets that exceed it. There are some ideas
floating around but determining the acceptable MTU all the way across
the internet is somewhat tricky.

Assuming you can choose any MTU you want, trunking looks more
attractive, the max allowed size of a UDP packet of 64K can contain
several hundred calls even at PCMU.  But this is unrealistic.  We are
most likely limited to the standard MTU in the neighborhood of 1500
and most guidelines suggest you only use a max percentage and you end
up with 1200 bytes per packet for payload data.  This only allows room
for trunking 7 PCMU calls.

Based on this conclusion it's obvious that only codecs that can
compress the audio better are even practical in trunking.  G.729 for
instance, can hold dozens of calls since its very compressed.   That
makes me feel to even bother making trunking, it should probably
revolve around some specific low bitrate codec.

Then there is a matter of implementation.  There are a few drafts on
how to do SIP/RTP trunking but none are formally adopted and new sip
drafts tend to be over engineered.  I've had some ideas on it but the
more I think of it, it pushes me towards making a dedicated protocol
for it, and if I bother with that, I may as well make a full blown
protocol that does everything else I always wanted from VoIP.

So every time I think about this issue i go in an endless circle and
end up just suggesting with Ken did and say use bigger ptimes between
the boxes in question.

On Mon, Jan 30, 2012 at 6:53 PM, Nowlin, Win <win at telisimo.com> wrote:
> Josue,
>
>> Is there any way to have a SIP Trunk. I mean to have for example 32
> channels merged in one? or something like this?
>>When i try to find SIP trunk on internet i just see options for TDM
> gateway or similar but not really a multiplexed trunk.
>
>        Let's define a SIP trunk for this discussion as "virtual
> internet connection" between your switch or gateway's IP address and
> your SIP provider's IP address.   Over this connection your SIP provider
> can send you any number of simultaneous conversations (basically
> equivalent to "channels" or "time slots" in the TDM world).  A SIP
> "trunk" can have as many simultaneous conversations ("Channels") as you
> wish, limited only by your internet bandwidth, how many "channels" you
> wish to pay for, and any limits set by your SIP provider.  Also the SIP
> trunk can have as many different DID numbers as you wish or as limited
> by your provider.  In this scenario, your provider is multiplexing the
> TDM sources into your SIP trunk.  In its most basic form, a SIP provider
> can "point" the SIP traffic directly at your Freeswitch's external IP
> address and there you are!!  Set up Freeswitch to be compatible with
> your Provider's requirements and you are there.
>
>        As far as TDM gateways are concerned, if you are using legacy
> equipment that requires TDM service, there are several SIP-TDM gateways
> available that can receive the SIP traffic from your Provider and
> convert it back-and-forth between SIP and TDM.  As far as the
> suitability and hardware requirements for Freeswitch to perform that
> function, since I am newly acquainted with Freeswitch, I leave that
> discussion to those who are experienced.
>
> Win N.
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Sergey Okhapkin
> Sent: Saturday, January 28, 2012 12:14
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] SIP Multiplexed Trunk as IAX2 or TDM
>
> SIP (and RTP) have no concept of trunking/audio frames multiplexing
> unlike
> IAX2 and TDM.
>
> On Saturday 28 January 2012 21:06:50 Josue Diaz Cruz wrote:
>> Is there any way to have a SIP Trunk. I mean to have for example 32
> channels
>> merged in one? or something like this? When i try to find SIP trunk on
>> internet i just see options for TDM gateway or similar but not really
> a
>> multiplexed trunk.
>>
>> Can we do something with freeswitch?
>>
>> Josue Diaz Cruz
>>
>> Departamento Tecnico y Soporte
>>
>>  <mailto:jdiaz at coinfru.com> jdiaz at coinfru.com
>>
>>
>>
>> C/ Balsicas 3
>>
>> Alquerias | 30580 | Murcia
>>
>>   <http://www.coinfru.com/> www.coinfru.com
>
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-- 
Anthony Minessale II

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