Re: [Asterisk-Users] Re: iax or sip
hide one end from the other. I have a customer and a carrier. I don't want one to know who the other is lest they get together and cut me out of the equation. This is certainly a valid point *right now* at a time when the industry is converting from PSTN to VoIP based transport. However, let's look a bit further down the road. For other services that have gone internet, changing the underlying technology has always come with a change in usage patterns and business models following suit. There is no reason to believe that telephony will be any different. Most of what defines SIP is clearly intended for a PSTN alike world. It is a reflection of the incumbents' wish that they can keep the status quo by piggy packing PSTN structures on top of TCP/IP. But is this how things are likely going to turn out? I tend to think not. Some of us have already got Asterisk running on sub $500 hardware running embedded Linux. It is only a matter of "when" not "if" that such devices will become commodity items that you will be able to pick up at Walmart or Radio Shack for about the same as a cordless phone base unit today. At that point, virtually every business and every household will have such a box. Couple that with some universal directory facility, ie ENUM and you have got a ubiquitous peer-to-peer telephone network where telcos will have no role to play other than providing the data pipes. I think everybody will agree that IAX is already perfectly suited for this kind of environment. So where does this leave SIP then? Most of the eeg-laying-wool-milk-sau features in SIP won't mean anything in this kind of environment since they are made for telco centric networks not for all peer-to-peer networks. As far as NAT and firewall traversal goes, SIP proponents are all too easy with offering workarounds that often introduce unwanted side-effects and they justify this by stating that NAT will go away with the advent of IPv6. I find that rather surprising. Let's face it, by the time that IPv6 is so widespread that NAT is not an issue anymore, there will be no telcos left to use all those SIP telco features because telephony will be an all peer-to-peer affair. And what space is left for value added services through third party providers can easily be accommodated by IAX. My conclusion is that IAX is far more future proof than any other VoIP protocol we know about today. Future proof in the sense of what the future is more likely to be, not in the sense of what the incumbents would like the future to be. just my 2 cents benjk __ Do You Yahoo!? http://bb.yahoo.co.jp/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: iax or sip
Randy Bush wrote: 1. Control a call, (maybe you want to do some ACL type filtering, maybe you want to keep track of usage, maybe you just to be in control...) Hmmm. Post setup, which clearly needs to go through all servers (or pbxen) in path, I don't see a win here. Send more clue. More likely is that the phones on the LAN are SIP and the boxes on the WAN are talking IAX, since that seems to make the most sense to me. hide one end from the other. I have a customer and a carrier. I don't want one to know who the other is lest they get together and cut me out of the equation. Yikes! Despite ad homina on this list, even I am not that sneaky. But I can see folk having legitimate needs such as this in an emerging market in desperate times. It's not always so sneaky... imagine that I'm a VoIP provider targeting homes and small business. I'm best to buy minutes in bulk from ATT and/or MCI or any carrier who can offer super-cheap rates in bulk. These carriers don't sell VoIP to home users, they sell it to people like me. I still don't really want my users or competition know where I buy my minutes from, nonetheless. And ATT doesn't really want a direct connection to the home user. So it works out that I am the logical middle-man, especially if I can trunk calls within a few hops of the user (like if I'm also their DSL provider as well) to save everybody bandwidth. Most significantly, this hierarchal paradigm is the most familiar to telcos and telephone people in general. Nick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: iax or sip
iax uses udp and traverses nats. neither of these seems useful to me. i loathe nats, and udp is not well-behaved in the sense of congestion avoidance. You may indeed loathe NATted networks, but in general they're very hard to avoid. Why would you criticize a protocol for dealing with such a thing efficiently--which, quite famously, SIP does not? i did not criticize the protocol. remember, my question started with i am looking at iax to see if it is applicable to my needs. i don't need nats, nat traversal, nat anything. if i did, iax might well be one of the technologies i would consider. but i don't. Do you know of a successful VoIP protocol that is entirely TCP-based? not currently, though folk are working hard on the congestion friendliness issue. if you're interested, i can point you to the relevant part of the ivtf basement. I would want the PBX in the datastream in cases where multiple endpoint connections would pass through multiple IAX boxen why? and yes, i mean the question. i see setup running through the boxen, of course. i just don't see why you would want the payload to traverse what might be a pretty baroque multi- continental path. i may have big pipes, but the bleedin' speed of light seems not to be very impressed. Perhaps in your case your networks are all public-IP, running on DS3s or OC48s. well not ds3s, stm-1 and above. but i ain't a big fan of wasting bytes. i am also not a fan of triangle routing. and maybe we could avoid the ad homina which seems to be too frequent on this list? randy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: iax or sip
On Mon, 5 Jul 2004, Randy Bush wrote: i did not criticize the protocol. remember, my question started with i am looking at iax to see if it is applicable to my needs. i don't need nats, nat traversal, nat anything. if i did, iax might well be one of the technologies i would consider. but i don't. Your up to no good again. Trying to troll? Look Bush it pretty easy - if you want a PBX use asterisk if not go away. regards joe ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: iax or sip
Okay, setting aside conspiracy theories, trolling, flaming, etc, let me summarize some differences between SIP and IAX, and it might help you make a decision about what is best for you. 1) IAX is more efficient on the wire than RTP for *any* number of calls, *any* codec. The benefit is anywhere from 2.4k for a single call to approximately trippling the number of calls per megabit for G.729 when measured to the MAC level when running trunk mode. 2) IAX is information-element encoded rather than ASCII encoded. This makes implementations substantially simpler and more robust to buffer overrun attacks since absolutely no text parsing or interpretation is required. The IAXy runs its entire IP stack, IAX stack, TDM interface, echo canceller, and callerid generation in 4k of heap and stack and 64k of flash. Clearly this demonstrates the implementation efficiency of its design. The size of IAX signalling packets is phenomenally smaller than those of SIP, but that is generally not a concern except with large numbers of clients frequently registering. Generally speaking, IAX2 is more efficient in its encoding, decoding and verifying information, and it would be extremely difficult for an author of an IAX implementation to somehow be incompatible with another implementation since so little is left to interpretation. 3) IAX has a very clear layer2 and layer3 separation, meaning that both signalling and audio have defined states, are robustly transmitted in a consistant fashion, and that when one end of the call abruptly disappears, the call WILL terminate in a timely fashion, even if no more signalling and/or audio is received. SIP does not have such a mechanism, and its reliability from a signalling perspective is obviously very poor and clumsy requiring additional standards beyond the core RF3261. 4) IAX's unified signalling and audio paths permit it to transparently navigate NAT's and provide a firewal administrator only a *single* port to have to open to permit its use. It requires an IAX client to know absolutely nothing about the network that it is on to operate. More clearly stated, there is *never* a situation that can be created with a firewall in which IAX can complete a call and not be able to pass audio (except of course if there was insufficient bandwidth). 5) IAX's authenticated transfer system allows you to transfer audio and call control off a server-in-the-middle in a robust fashion such that if the two endpoints cannot see one another for any reason, the call continues through the central server. 6) IAX clearly separates Caller*ID from the authentication mechanism of the user. SIP does not have a clear method to do this unless Remote-Party-ID is used. 7) SIP is an IETF standard. While there is some fledgling documentation courtesy Frank Miller, IAX is not a published standard at this time. 8) IAX allows an endpoint to check the validity of a phone number to know whether the number is complete, may be complete, or is complete but could be longer. There is no way to completely support this in SIP. 9) IAX always sends DTMF out of band so there is never any confusion about what method is used. 10) IAX support transmission of language and context, which are useful in an Asterisk environment. That's pretty much all that comes to mind at the moment. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: iax or sip
Great piece of Info Mark, THANK YOU...very educational to me at least. Do you perhaps have more of these gems somewhere where I can peruse them at a time I can devote some time to my continuing education on * and IAX? It would be a very valuable resource to many of us who are still on the steep part of the learning curve. Thanks in advance. Marc At 19:59 7/5/2004, you wrote: Okay, setting aside conspiracy theories, trolling, flaming, etc, let me summarize some differences between SIP and IAX, and it might help you make a decision about what is best for you. 1) IAX is more efficient on the wire than RTP for *any* number of calls, *any* codec. The benefit is anywhere from 2.4k for a single call to approximately trippling the number of calls per megabit for G.729 when measured to the MAC level when running trunk mode. 2) IAX is information-element encoded rather than ASCII encoded. This makes implementations substantially simpler and more robust to buffer overrun attacks since absolutely no text parsing or interpretation is required. The IAXy runs its entire IP stack, IAX stack, TDM interface, echo canceller, and callerid generation in 4k of heap and stack and 64k of flash. Clearly this demonstrates the implementation efficiency of its design. The size of IAX signalling packets is phenomenally smaller than those of SIP, but that is generally not a concern except with large numbers of clients frequently registering. Generally speaking, IAX2 is more efficient in its encoding, decoding and verifying information, and it would be extremely difficult for an author of an IAX implementation to somehow be incompatible with another implementation since so little is left to interpretation. 3) IAX has a very clear layer2 and layer3 separation, meaning that both signalling and audio have defined states, are robustly transmitted in a consistant fashion, and that when one end of the call abruptly disappears, the call WILL terminate in a timely fashion, even if no more signalling and/or audio is received. SIP does not have such a mechanism, and its reliability from a signalling perspective is obviously very poor and clumsy requiring additional standards beyond the core RF3261. 4) IAX's unified signalling and audio paths permit it to transparently navigate NAT's and provide a firewal administrator only a *single* port to have to open to permit its use. It requires an IAX client to know absolutely nothing about the network that it is on to operate. More clearly stated, there is *never* a situation that can be created with a firewall in which IAX can complete a call and not be able to pass audio (except of course if there was insufficient bandwidth). 5) IAX's authenticated transfer system allows you to transfer audio and call control off a server-in-the-middle in a robust fashion such that if the two endpoints cannot see one another for any reason, the call continues through the central server. 6) IAX clearly separates Caller*ID from the authentication mechanism of the user. SIP does not have a clear method to do this unless Remote-Party-ID is used. 7) SIP is an IETF standard. While there is some fledgling documentation courtesy Frank Miller, IAX is not a published standard at this time. 8) IAX allows an endpoint to check the validity of a phone number to know whether the number is complete, may be complete, or is complete but could be longer. There is no way to completely support this in SIP. 9) IAX always sends DTMF out of band so there is never any confusion about what method is used. 10) IAX support transmission of language and context, which are useful in an Asterisk environment. That's pretty much all that comes to mind at the moment. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: iax or sip
There are many reasons to have an Asterisk box in a stream: 1. Control a call, (maybe you want to do some ACL type filtering, maybe you want to keep track of usage, maybe you just to be in control...) hmmm. post setup, which clearly needs to go through all servers (or pbxen) in path, i don't see a win here. send more clue. 2. Provide features (access to PSTN, conference capability, music on hold, call parking, agents and queues. the list goes on and on) that's setup not payload 3. Endpoints (User Agents) MAY not be able to send data streams to each other directly (firewalls or nats in the middle) yes indeed. so one, but likely only one, if they're asterisk, pbx needs to intermediate, not a bunch on a path. And depending upon your view of things (your view might be different than the view of the IT/communications administrator of a large company), using IAX in a geographically distributed use scenario might very well be exactly what you want (use over an encrypted vpn link, etc.) yes, it might be. but as you know, i am a big pipe backbone geek, not an admin of a large distributed company. and i am addict of simple (non-complex, not the presence protocol:-). randy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: iax or sip
1. Control a call, (maybe you want to do some ACL type filtering, maybe you want to keep track of usage, maybe you just to be in control...) hmmm. post setup, which clearly needs to go through all servers (or pbxen) in path, i don't see a win here. send more clue. hide one end from the other. I have a customer and a carrier. I don't want one to know who the other is lest they get together and cut me out of the equation. yikes! despite ad homina on this list, even i am not that sneaky. but i can see folk having legitimate needs such as this in an emerging market in desperate times. My comment above not withstanding, might I be correct that your purpose is more along the lines of a personal comm system? while i have that going on the side for fun, see appended, the use for which i am scratching my head is big pipe global backbone stuff. i am in the commercial world in my daytime job. i sold my soul long ago; had to put kids through college and all that crass capitalist stuff. randy --- for your amusement, i can talk about my private play-pen i have a rack in seattle's carrier hotel with 2xSTM-1 connectivity. in it, i have o asterisk running on a freebsd server (many many thanks to the freebsd porting crew) o cisco 1750 with pots out-dial on fixed ld price plan in our home nearby on bainbridge island, we have o cisco 7960 on an external static address o spa-3000 on an external static address - one port to in-house phone system - pstn port to local telco, qwest in our home on the big island of hawaii, we have the same as bainbridge o cisco 7960 on an external static address o spa-3000 on an external static address - one port to in-house phone system - pstn port to local telco, verizon we use the system to get o follow-me forwarding to wherever we are, including mobile phones, blah blah, so that callers don't have to know where we are to call us o free calls between the two houses o low cost calls within north america o gate to low cost voip intl pstn gateway provider, as we make a lot of personal intl calls fairly simple and boring. and thanks to a number of folk who helped me up the learning curve (sjw being the first, and i am not even paying his counselling [sic] bill:-). the telco part of this stuff is not easy for an over-attenuated ip kinda guy. and my programming language background does me no good with asterisk config files! :-) -30- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
