Re: [asterisk-users] IAX vs SIP
Hi, Just to review the test I did: ---SIP extension-- Trunk - | SIPp |---| Asterisk 1 || Asterisk 2 | ------ -- Both Asterisk boxes are virtual machines in VirtualBox and version 1.4.21.2. I generated calls using SIPp, and I monitored the cpu utilization in the Asterisk 1 with top. I compare the cpu utilization when I used IAX and when I used SIP as Trunk protocol. The following are the results (averages) I got with ulaw codecs in both sides: Calls IAX SIP 4 6,0 1,8 10 9,2 4,6 20 19,58,6 30 28,213,5 34 36,916,2 40 38,219,5 50 36,924,3 As you can see, IAX almost doubles the cpu cycles. I repeated the test using gsm as the trunk codec, and in this scenario IAX shows a better performance (Sip extension continues with ulaw): IAX SIP 1,8 25,7 12,841,5 29,247,0 45,769,4 54,278,5 53,383,3 65,787,1 And finally I repeated with iLBC in the trunk, and SIP won again: IAX SIP 8,5 9,4 29,314,5 57,623,6 74,437,3 84,341,5 -- 51,2 -- 67,0 Does this makes sense? Any feedback? Has anybody done similar test for comparison? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX vs SIP
On 7 Sep 2008, at 21:34, Edgar Guadamuz wrote: Hello, I have been testing a trunk IAX and another SIP, using sipp to generate SIP calls to a Asterisk box. The testing dialplan just connects to another Asterisk box, who answers the call and playback some files. I noticed that the cpu load is higher when I use an IAX, about 90% for 25 simultaneous calls. In the other hand, with a SIP trunk the cpu load was about the half or less. In both cases the Asterisk box was in the middle of the RTP path, and both the trunk and the sip client had the same codec, ulaw. Does it make sense? Why is IAX demanding so much cpu load? Which Asterisk version are you running? There was a specific version (1.4.20 I think) that had made IAX super-expensive. The most recent versions of asterisk _should_ have IAX being roughly equivalent in CPU usage as SIP Incidentally if anyone has comparative numbers for IAX vs SIP on 1.6 betas (or hyper-recent 1.4) I'd love to have them for a talk I'm doing at astricon. Tim. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX vs SIP
Hello, I have been testing a trunk IAX and another SIP, using sipp to generate SIP calls to a Asterisk box. The testing dialplan just connects to another Asterisk box, who answers the call and playback some files. I noticed that the cpu load is higher when I use an IAX, about 90% for 25 simultaneous calls. In the other hand, with a SIP trunk the cpu load was about the half or less. In both cases the Asterisk box was in the middle of the RTP path, and both the trunk and the sip client had the same codec, ulaw. Does it make sense? Why is IAX demanding so much cpu load? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX vs SIP - Getting Asterisk out of the media path
If a call comes into my Asterisk server on a DiD provided by an ITSP and the dialplan sends that call to another external number throught the same ITSP's network, I don't want the RTP packets to pass through my server once the call is bridged. I have had great success getting this to work using IAX, but I have not been able to get this to work with SIP. The call is bridged OK (media at both ends) but the media continues passing through my network. The default behaviour for the Dial command is to have Asterisk step out of the media path provided you avoid some options like tT, which I do, so this should work. One interesting note: In an Ethereal trace, I see 407 Proxy Authentication required just after the INVITE to the callee. Could that be part of the problem? If so what's the fix? I thought it had something to do with the auth parameter. I am: - Behind a NAT, - Running Red Hat 9.0 - Running Asterisk 1.2.14 How do I stop the media passsing through my Asterisk server after a call between two external parties has been bridged? My sip.conf and the dial command I use are below. Thanks, Hugh ;*** Dial Command *** exten = _6136930630,n,Dial(SIP/[EMAIL PROTECTED]) ; SIP.conf ** [general] ; context=incoming-bogus-calls bindport=5060 bindaddr=0.0.0.0 maxexpirey=3600 defaultexpirey=3600 notifymimetype=text/plain rtptimeout=60 rtpholdtimeout=300 disallow=all allow=ulaw ; ; This section is because i'm behind nat ; externip=999.99.999.99 ;Outside address localnet=192.168.0.148/255.255.255.0 ;Inside Network ; register=6135551234:[EMAIL PROTECTED]/6135551234 ; [6135551234] type=peer ;auth=md5 auth=6135551234:[EMAIL PROTECTED] username=6135551234 fromuser=6135551234 fromdomain=myITSP.ca secret= host=sip02.myITSP.ca port=5060 nat=yes canreinvite=yes qualify=no disallow=all allow=ulaw dtmfmode=rfc2833 insecure=very context=incoming-sip -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.411 / Virus Database: 268.17.36/681 - Release Date: 11/02/2007 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX vs SIP trunks between Asterisk boxes
Brad Templeton wrote: On Sun, Jan 07, 2007 at 04:12:27PM +, Thomas Kenyon wrote: Brad Templeton wrote: For SIP phone calling * box, relay to other * box and out to SIP phone, you definitely want SIP all the way. Unless bandwidth between the * servers is a concern, then you're better off keeping the link between servers as IAX. (preferably trunked) The bandwidth of the audio stream dwarfs the bandwidth of signalling traffic by orders of mangitude. So in fact, I think this is exactly wrong. If bandwidth to or between the servers is a concern, that's where you most want to not be in the audio path. But if you have multiple RTP streams emnbedded in an IAX trunk, then the IP overhead is significantly reduced. AFAIK video should work for IAX2, there is explicit support for it. (unlike h323). Asterisk will only be able to pass the raw RTP traffic though, since it doesn't have any video codecs (just format definitions). ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX vs SIP trunks between Asterisk boxes
Brad Templeton wrote: For SIP phone calling * box, relay to other * box and out to SIP phone, you definitely want SIP all the way. Unless bandwidth between the * servers is a concern, then you're better off keeping the link between servers as IAX. (preferably trunked) It is worth remembering in this sort of setup, often the phones at one site will not have a route to the phons on the other site, so the calls wont be re-invited off to the handsets anyway. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX vs SIP trunks between Asterisk boxes
Unless bandwidth between the * servers is a concern, then you're better off keeping the link between servers as IAX. (preferably trunked) As I understand it video will NOT work if you use an IAX trunk between * boxes, it must be SIP. Just food for thought in case you are planning on using video. David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX vs SIP trunks between Asterisk boxes
On Sun, Jan 07, 2007 at 04:12:27PM +, Thomas Kenyon wrote: Brad Templeton wrote: For SIP phone calling * box, relay to other * box and out to SIP phone, you definitely want SIP all the way. Unless bandwidth between the * servers is a concern, then you're better off keeping the link between servers as IAX. (preferably trunked) The bandwidth of the audio stream dwarfs the bandwidth of signalling traffic by orders of mangitude. So in fact, I think this is exactly wrong. If bandwidth to or between the servers is a concern, that's where you most want to not be in the audio path. It is worth remembering in this sort of setup, often the phones at one site will not have a route to the phons on the other site, so the calls wont be re-invited off to the handsets anyway. If it's phone-on-NAT to phone-on-different-NAT, it typically will not work. That doesn't mean it can't work if bandwidth is important. I think the complete solution, not yet in Asterisk as I understand it is for Asterisk to be aware of both the internal and external addresses of a phone, and to connect internal phones with their internal addresses, but to connect internal phones to external endpoints through their external addresses. Ideally audio never flows through asterisk unless it's doing an IVR dialogue or otherwise explicitly wants it to. (In fact, ideally DTMF goes via SIP INFO or its successors so that Asterisk can listen to the DTMF without being in on the audio.) Flowing audio through your box costs not just bandwidth, it adds latency, and very slight extra risks of packet loss. Latency is the bane of voip calls, it also worsens echo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX vs SIP trunks between Asterisk boxes
On Fri, Jan 05, 2007 at 11:33:02AM +, Gordon Henderson wrote:
On Thu, 4 Jan 2007, Noah Miller wrote:
Hi Damon -
Can anyone comment on the overhead added when a SIP call comes into one
asterisk box, is routed to another with IAX instead of SIP, and is then
sent
to the UA from the second box with SIP?
DTMF passthrough issues?
I've got a client with sip phones on several different servers and
IAX links between the servers, so I guess that's pretty similar to
your setup. I've never bothered to check for overhead since it was
never an issue (all servers are P4 2.8 ghz or Xeon 2.8 ghz, 1GB ram,
with never more than 3-4 calls going through any one of the IAX
links). I can say that DTMF works fine in this setup.
I'm doing the same on 1GHz processors - CPU usage is virtually nil unless
there's transcoding going on (about 4% per GSM transcode)
ADSL bandwidth is more of a concern for me in these applications )-:
While it would be work to set up, you actually ideally want to
trunk with the same protocol being used by the external phones
or endpoints. When connecting a SIP to SIP call (presuming you
don't have annoying nat problems or have turned canreinvite off)
the audio should go directly from endpoint to endpoint and not
via asterisk.Ditto on IAX to IAX calls.
For SIP phone calling * box, relay to other * box and out to SIP
phone, you definitely want SIP all the way.
In some ways, an ideal solution would have two trunk connections
between the boxes (really just two config entries in iax.conf and
sip.conf) and go between the boxes with whatever protocol the
calling channel is using. You could write dialplan scripts to
pull out the channel and choose the right * to * protocol (as
opposed to inter-asterisk protocol which has another meaning.
:-)
It can also be worth having a termination provider that you
can talk to with both IAX and SIP, and sending them the call
with the same protocol the phone used.
Annoyingly, IAX and SIP channels use different interfaces
to provide the address, so you can't do
DIAL(${chantype}/[EMAIL PROTECTED])
A cute patch would be to support that with a consistent syntax over
channels.
Note if you use various flags on Dial which require asterisk
to hear dtmf or do other audio, you are stuck hairpinning.
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Re: [asterisk-users] IAX vs SIP trunks between Asterisk boxes
On Thu, 4 Jan 2007, Noah Miller wrote: Hi Damon - Can anyone comment on the overhead added when a SIP call comes into one asterisk box, is routed to another with IAX instead of SIP, and is then sent to the UA from the second box with SIP? DTMF passthrough issues? I've got a client with sip phones on several different servers and IAX links between the servers, so I guess that's pretty similar to your setup. I've never bothered to check for overhead since it was never an issue (all servers are P4 2.8 ghz or Xeon 2.8 ghz, 1GB ram, with never more than 3-4 calls going through any one of the IAX links). I can say that DTMF works fine in this setup. I'm doing the same on 1GHz processors - CPU usage is virtually nil unless there's transcoding going on (about 4% per GSM transcode) ADSL bandwidth is more of a concern for me in these applications )-: Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX vs SIP trunks between Asterisk boxes
In order to work around some authentication issues I am considering connecting two asterisk boxes with IAX instead of SIP. The original reason for choosing SIP was to reduce the need to translate SIP signaling to IAX, since all origination, termination, and UAs are SIP. Can anyone comment on the overhead added when a SIP call comes into one asterisk box, is routed to another with IAX instead of SIP, and is then sent to the UA from the second box with SIP? DTMF passthrough issues? Any other issues? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX vs SIP trunks between Asterisk boxes
Hi Damon - Can anyone comment on the overhead added when a SIP call comes into one asterisk box, is routed to another with IAX instead of SIP, and is then sent to the UA from the second box with SIP? DTMF passthrough issues? I've got a client with sip phones on several different servers and IAX links between the servers, so I guess that's pretty similar to your setup. I've never bothered to check for overhead since it was never an issue (all servers are P4 2.8 ghz or Xeon 2.8 ghz, 1GB ram, with never more than 3-4 calls going through any one of the IAX links). I can say that DTMF works fine in this setup. - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] iax vs. sip?
Hi Steven,The provider's implementation will have a bigger affect than any differences within Asterisk, e.g. how they are load-balancing and whether in fact SIP is serviced by Asterisk at all. Compared like-for-like within Asterisk we find there is not a lot in it, with each having their own pros and cons. We support both and whilst we have more customers on SIP than IAX, currently favour IAX for new customers where they are undecided given lower support overhead and simplified load-balancing. I'd recommend you try both with the provider you're considering. Simonwww.esms.comOn 8/31/06, BerkHolz, Steven [EMAIL PROTECTED] wrote: I have no NAT issues. My PBX is multihomed and the outside IP is locked down for all except IAX and SIP ports. With the current version of asterisk, which transport is better right now? I am looking at 6-10 simultaneous calls over a half T1. I am not asking about codecs here, I am asking about SIP vs. IAX if the provider does either. (we are looking at testing Teliax next) I have seen posts about jitter in IAX, so I am not sure if SIPmight bebetter to use right now. Also, since IAX uses the same port for all of the calls, the call separation has to be done higher in the OSI stack. I do not know if this is better or worse or neither. Thank You, Steven BerkHolz- MCSA - MCSE -Manager of Information SystemsTESCO Group CompaniesFax. 248-836-5101www.TESCOGroup.com Board member ofwww.glimasoutheast.org ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] iax vs. sip?
We've been using iax with teliax.com for a couple of years, and it seems the quality of calls varies with time. Sometimes it is good and next time its not so good. There has been changes occurring to iax and the jitterbuffer stuff over the last two years, and I'm reasonably certain that some poor quality is related to differences between teliax.com's implementation (eg, s/w versions) and ours. I've not bother to try sip since our asterisk implementation is truly both a production box for our small office, and a test box for various version testing, etc. We used iax for more than a year and moved to sip about 6 months ago. The quality from termination providers seems much better now with sip. Tom At 09:38 PM 8/30/2006, you wrote: I have no NAT issues. My PBX is multihomed and the outside IP is locked down for all except IAX and SIP ports. With the current version of asterisk, which transport is better right now? I am looking at 6-10 simultaneous calls over a half T1. I am not asking about codecs here, I am asking about SIP vs. IAX if the provider does either. (we are looking at testing Teliax next) I have seen posts about jitter in IAX, so I am not sure if SIP might be better to use right now. Also, since IAX uses the same port for all of the calls, the call separation has to be done higher in the OSI stack. I do not know if this is better or worse or neither. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] iax vs. sip?
I have no NAT issues. My PBX is multihomed and the outside IP is locked down for all except IAX and SIP ports. With the current version of asterisk, which transport is better right now? I am looking at 6-10 simultaneous calls over a half T1. I am not asking about codecs here, I am asking about SIP vs. IAX if the provider does either. (we are looking at testing Teliax next) I have seen posts about jitter in IAX, so I am not sure if SIPmight bebetter to use right now. Also, since IAX uses the same port for all of the calls, the call separation has to be done higher in the OSI stack. I do not know if this is better or worse or neither. Thank You, Steven BerkHolz- MCSA - MCSE -Manager of Information SystemsTESCO Group CompaniesFax. 248-836-5101www.TESCOGroup.com Board member ofwww.glimasoutheast.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] iax vs. sip?
We used iax for more than a year and moved to sip about 6 months ago. The quality from termination providers seems much better now with sip. Tom At 09:38 PM 8/30/2006, you wrote: Content-class: urn:content-classes:message Content-Type: multipart/alternative; boundary=_=_NextPart_001_01C6CCA6.8EFA1438 I have no NAT issues. My PBX is multihomed and the outside IP is locked down for all except IAX and SIP ports. With the current version of asterisk, which transport is better right now? I am looking at 6-10 simultaneous calls over a half T1. I am not asking about codecs here, I am asking about SIP vs. IAX if the provider does either. (we are looking at testing Teliax next) I have seen posts about jitter in IAX, so I am not sure if SIP might be better to use right now. Also, since IAX uses the same port for all of the calls, the call separation has to be done higher in the OSI stack. I do not know if this is better or worse or neither. Thank You, Steven BerkHolz - MCSA - MCSE - Manager of Information Systems TESCO Group Companies Fax. 248-836-5101 www.TESCOGroup.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX vs SIP (music on hold)
Does IAX support music on hold? It seems only my SIP phones do. Is this correct? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] IAX vs SIP (music on hold)
I hope I didnt get your question wrong, but if you are asking whether Asterisk can play MOH to an IAX client, then the answer is yes. We have a couple of IAX clients connecting into the queue and are being played MOH while waiting for an operations. Hope this helps :) Cheers On Tue, 29 Mar 2005 14:49:26 -0500, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Does IAX support music on hold? It seems only my SIP phones do. Is this correct? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] IAX vs SIP (music on hold)
[EMAIL PROTECTED] wrote: Does IAX support music on hold? It seems only my SIP phones do. Is this correct? As I understand it, once the call is delivered to asterisk, it becomes abstracted into a channel. And you can do anything to one channel that you can do to other channels ( with a few notable exceptions including zap channels ). So it shouldn't make a difference whether it's sip/iax/zap as far as MoH is concerned. What may cause issues is what class of MoH is specified, by default and otherwise. But as I haven't tinkered with that a great deal yet, I can't tell you much beyond that. Good luck Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] IAX vs SIP
Thanks, this is exactly what I was looking for. I tried experimenting with different codecs myself, and GSM seems to be the only one that works... neither iLBC or Speex went thru. I'm using XLite v1.x Asterisk 0.5.0, wonder if it's a softphone's problem? I have got X-Lite to work with G.711 and GSM only, I have never been able to get it to work with iLBC or Speex.. I use iLBC over my IAX trunk and it works fine so I can only guess that there is some compatibility problem between X-Lite and Asterisk.. Later -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX vs SIP
Does this thread help? http://lists.digium.com/pipermail/asterisk-users/2003-June/014788.html Thanks, this is exactly what I was looking for. I tried experimenting with different codecs myself, and GSM seems to be the only one that works... neither iLBC or Speex went thru. I'm using XLite v1.x Asterisk 0.5.0, wonder if it's a softphone's problem? Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX vs SIP
How do you set up IAX in Trunk mode? Uriel Add trunk=yes to your definition in iax.conf.. Later -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX vs SIP
FYI: trunking only works in IAX2 and it requires you to have a zaptel interface on both endpoints I have heard that but in my setup I only have Zaptel hardware on one side and trunking appears to work fine.. Initially I tried using ztdummy on the side which didn't have zaptel hardware but this caused the trunk to break properly, without it it works fine.. Maybe I just have a freak setup.. :) Later.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX vs SIP
Hello, On 19-09 19:48, WipeOut . wrote: Also IAX does not care about NAT so a situation like.. AST--NAT--Internet--NAT--AST ..will work fine.. SIP will have problems in a setup like this without the use of specialised NAT routers.. I am wondering how setup like this could work with IAX (or any other protocol) when symmetric NATs are used. If you have two different NATs then direct connection is not possible between hosts behind those two NATs. You have to do some kind of provisioning of the NAT boxes (i.e. port forwarding). Jan. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX vs SIP
I am wondering how setup like this could work with IAX (or any other protocol) when symmetric NATs are used. If you have two different NATs then direct connection is not possible between hosts behind those two NATs. You have to do some kind of provisioning of the NAT boxes (i.e. port forwarding). Jan. You setup port forwarding on your each NAT's to the server behind the NAT.. If you don't have a static IP or resolvable DNS name on one of the boxes you can get it to register with the remote side.. You will have to have the NAT's public IP on at least one side static or resolvable through some form of DNS or DDNS.. Later.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX vs SIP
I wonder how IAX compares to SIP bandwidth-wise? I've tried both over overseas IP connection, and somehow SIP seemed to work better. Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX vs SIP
I wonder how IAX compares to SIP bandwidth-wise? I've tried both over overseas IP connection, and somehow SIP seemed to work better. Peter Then try making two or three or more calls at the same time.. :) If you setup IAX in trunk mode it uses the same connection for multiple voice streams and so optimises the bandwith usage by reducing the overhead per voice channel.. SIP can't do that.. Also IAX does not care about NAT so a situation like.. AST--NAT--Internet--NAT--AST ..will work fine.. SIP will have problems in a setup like this without the use of specialised NAT routers.. Later.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX vs SIP
Does this thread help? http://lists.digium.com/pipermail/asterisk-users/2003-June/014788.html On Fri, Sep 19, 2003 at 01:18:53PM -0500, Peter Zeltins wrote: I wonder how IAX compares to SIP bandwidth-wise? I've tried both over overseas IP connection, and somehow SIP seemed to work better. Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX vs SIP
On Fri, 19 Sep 2003, WipeOut . wrote: I wonder how IAX compares to SIP bandwidth-wise? I've tried both over overseas IP connection, and somehow SIP seemed to work better. Peter Then try making two or three or more calls at the same time.. :) If you setup IAX in trunk mode it uses the same connection for multiple voice streams and so optimises the bandwith usage by reducing the overhead per voice channel.. SIP can't do that.. Also IAX does not care about NAT so a situation like.. AST--NAT--Internet--NAT--AST ..will work fine.. SIP will have problems in a setup like this without the use of specialised NAT routers.. FYI: trunking only works in IAX2 and it requires you to have a zaptel interface on both endpoints James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX vs SIP
How do you set up IAX in Trunk mode? Uriel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of WipeOut . Sent: Friday, September 19, 2003 3:49 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] IAX vs SIP I wonder how IAX compares to SIP bandwidth-wise? I've tried both over overseas IP connection, and somehow SIP seemed to work better. Peter Then try making two or three or more calls at the same time.. :) If you setup IAX in trunk mode it uses the same connection for multiple voice streams and so optimises the bandwith usage by reducing the overhead per voice channel.. SIP can't do that.. Also IAX does not care about NAT so a situation like.. AST--NAT--Internet--NAT--AST ..will work fine.. SIP will have problems in a setup like this without the use of specialised NAT routers.. Later.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
