SIP Multiplexed Trunk as IAX2 or TDM
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
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
SIP only has gateways, no trunks. When you're dealing with VoIP, treat SIP like data, not like voice channels. On Sat, Jan 28, 2012 at 12:18 PM, <georg@riseup.net> wrote:
Hi,
SIP (and RTP) have no concept of trunking/audio frames multiplexing unlike IAX2 and TDM.
Are you sure with this? I've got a sip trunk trough a german provider with eight channels.
Greetings, Georg
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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It does make a considerable difference if you are running your RTP through IP encryption (eg IPSec VPN) then the per-packet overhead becomes a significant factor. Companies such as NET (www.net.com) provide switches (VX400, VX900, etc) that run bespoke SIP-like and RTP-like protocols that merge RTP calls between 2 WAN nodes into single packets and this can reduce the per-packet overhead of encryption quite significantly. The cost is a small additional latency (eg 20-40ms) while the packets from various calls are aggregated. John PS. I have no affiliation with NET. On 28/01/12 20:26, georg@riseup.net wrote:
SIP only has gateways, no trunks. When you're dealing with VoIP, treat SIP like data, not like voice channels. Allright, but makes this a difference in practice?
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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There is no such thing as trunking in sip as in trunking like iax2... There are much easier ways to deal with this in sip... For instance set your ptime between freeswitch boxes to be 40ms, this effectively cuts the RTP/PPS overhead in 1/2... Set it to 60ms cut it by 66%.... This works by increasing the amount of audio sent in 1 pack from 20ms to 40 or 60ms... Merging RTP between calls after that point is pretty much a mute point as you already are achieving a large reduction in RTP overhead... K On 1/28/12 2:42 PM, "John" <freeswitch@earthspike.net> wrote:
It does make a considerable difference if you are running your RTP through IP encryption (eg IPSec VPN) then the per-packet overhead becomes a significant factor. Companies such as NET (www.net.com) provide switches (VX400, VX900, etc) that run bespoke SIP-like and RTP-like protocols that merge RTP calls between 2 WAN nodes into single packets and this can reduce the per-packet overhead of encryption quite significantly. The cost is a small additional latency (eg 20-40ms) while the packets from various calls are aggregated.
John
PS. I have no affiliation with NET.
On 28/01/12 20:26, georg@riseup.net wrote:
SIP only has gateways, no trunks. When you're dealing with VoIP, treat SIP like data, not like voice channels. Allright, but makes this a difference in practice?
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
"Channels" and "audio frames multiplexing" to preserve bandwidth are different things. On Saturday 28 January 2012 21:18:31 georg@riseup.net wrote:
Hi,
SIP (and RTP) have no concept of trunking/audio frames multiplexing unlike IAX2 and TDM.
Are you sure with this? I've got a sip trunk trough a german provider with eight channels.
Greetings, Georg
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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
________________________________________________________________________ _ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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
________________________________________________________________________ _ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
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
________________________________________________________________________ _ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Kerem Erciyes - Sistem Danismani http://keremerciyes.com
But back to the main point of your question, yes, you should be able to find many providers who will offer you quite a few simultaneous outbound or inbound channels. There are too many to list so I'll just point you to VoIP-Info: http://www.voip-info.org/wiki/view/VOIP+Service+Providers+B2B Nicholas Blasgen Developer Predictive Dialer Limited +1 (724) 252-7436 (cell) Skype: nblasgen www.hellohunter.com: Providing wholesale and white labeled predictive dialing and voice broadcasting services. On Sat, Jan 28, 2012 at 12:06 PM, Josue Diaz Cruz <jdiaz@coinfru.com> 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?
participants (10)
-
Anthony Minessale
-
curriegrad2004
-
georg＠riseup.net
-
John
-
Josue Diaz Cruz
-
Ken Rice
-
Kerem Erciyes
-
Nicholas Blasgen
-
Nowlin, Win
-
Sergey Okhapkin