I used IAX2 trunks between Asterisk boxes back when I did VoIP with Asterisk and even though it was bandwidth efficient on paper, whenever the MPLS backbone (with Voice networks prioritised mind you) was a little congested we experienced stutter in all cals made on the trunk and only fix was to hang-up and dial again. When we switched to SIP stutters and problems still happened sometimes, but not all calls failed at the same time, which made users believe the PBX was not broken after all. Just another of my horrible experiences debugging VoIP problems and running on minimal infrastructure. Regards, Kerem On Tue, Jan 31, 2012 at 7:36 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
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@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@lists.freeswitch.org [mailto:freeswitch-users-bounces@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@coinfru.com> jdiaz@coinfru.com
C/ Balsicas 3
Alquerias | 30580 | Murcia
<http://www.coinfru.com/> www.coinfru.com
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