Simple IAX2 setup - help with converting from asterisk to freeswitch
Hi, I have compiled freeswitch trunk from svn on debian. I am trying to convert the following simple working asterisk config to freeswitch and I would be really grateful if someone could point me in the right direction: pbx:/etc/asterisk# cat iax.conf [general] bindport=4569 bindaddr=my.ip.address jitterbuffer=yes disallow=all allow=ulaw allow=alaw context=deadend [voiptalk] type=peer username=XXXXXX secret=XXXXXX host=iax.voiptalk.org [0843XXXXXX] type=friend username=08433XXXXXX context=incoming requirecalltoken=auto [1000] type=friend host=dynamic mailbox=1000 secret=XXXXXX context=phones requirecalltoken=auto pbx:/etc/asterisk# cat extensions.conf [general] autofallthrough=yes [outgoing] exten => _0[1-9].,1,Dial(IAX2/XXXXXX@voiptalk/44${EXTEN:1}) exten => _00.,1,Dial(IAX2/XXXXXX@voiptalk/${EXTEN:2}) [internal] exten => 901,1,VoiceMailMain() exten => 901,2,Hangup() exten => 902,1,MeetMe(1234,cdM) exten => 902,2,Hangup() [incoming] exten => XXXXXXXX,1,Dial(IAX2/1000,30) exten => XXXXXXXX,2,VoiceMail(1000@internal) [phones] include => internal include => outgoing pbx:/etc/asterisk# cat voicemail.conf [internal] 1000 => XXXX,Matt,me@myomain.com format=wav49 maxsilence=0 Sorry to be a bonehead, but I'm struggling with the wiki docs (especially with regard to IAX2) and also concerned about security. I installed the samples when I compiled freeswitch but wondering if that is a security risk as the PBX box is on the public internet? Many thanks, Matt
iax2 support has been removed from FreeSWITCH in current trunk and will not be in the 1.0.5 release. On Feb 4, 2010, at 7:22 AM, Matthew Law wrote:
Hi,
I have compiled freeswitch trunk from svn on debian. I am trying to convert the following simple working asterisk config to freeswitch and I would be really grateful if someone could point me in the right direction:
<snip>
Sorry to be a bonehead, but I'm struggling with the wiki docs (especially with regard to IAX2) and also concerned about security. I installed the samples when I compiled freeswitch but wondering if that is a security risk as the PBX box is on the public internet?
Mike
Why is that? - a lot of web pages I have read claim that IAX is more secure and efficient. I have no problem with using SIP whatsoever and it certainly appears to be ubiquitous. I am a complete newcomer to VoIP and I am trying to do this as securely as possible since I want to run freeswitch on a Xen VPS which will be visible on the internet. Anyway, since my first question, I have worked my way through the wiki, read a lot of example configs and made some sense of the docs. I now have a very basic config working (with SIP) on a local vmware image using the 'quick and dirty' Makefile method. I removed all of the example configs from the xml files (those default extensions and passwords scared me) and added just one for extension 1000, plus my dialplan and inbound/outbound settings. One question: is there any reason not to use a smaller extension number range, like 200-210, for example? Now to figure out how to get time based roaming working... Thanks, Matt. On Fri, February 5, 2010 6:43 am, Michael Jerris wrote:
iax2 support has been removed from FreeSWITCH in current trunk and will not be in the 1.0.5 release.
Mike
the lib that we used to provide iax support is pretty much abandonware (no longer updated) and newer iax implementations (like latest asterisk) can cause it to crash. There are no license compatible iax implementations that work, so.. mod_iax has been moved to the unsupported column. Default passwords -- that is a single var in vars.xml that controls the passwords. number ranges - up to you. The sample configs supplied are just that, samples. I use a smaller range personally. On Fri, Feb 5, 2010 at 7:53 AM, Matthew Law <matt@webcontracts.co.uk> wrote:
Why is that? - a lot of web pages I have read claim that IAX is more secure and efficient. I have no problem with using SIP whatsoever and it certainly appears to be ubiquitous. I am a complete newcomer to VoIP and I am trying to do this as securely as possible since I want to run freeswitch on a Xen VPS which will be visible on the internet.
Anyway, since my first question, I have worked my way through the wiki, read a lot of example configs and made some sense of the docs. I now have a very basic config working (with SIP) on a local vmware image using the 'quick and dirty' Makefile method. I removed all of the example configs from the xml files (those default extensions and passwords scared me) and added just one for extension 1000, plus my dialplan and inbound/outbound settings.
One question: is there any reason not to use a smaller extension number range, like 200-210, for example?
Now to figure out how to get time based roaming working...
Thanks,
Matt.
On Fri, February 5, 2010 6:43 am, Michael Jerris wrote:
iax2 support has been removed from FreeSWITCH in current trunk and will not be in the 1.0.5 release.
Mike
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There's a fairly simple solution to IAX needs, which is to run Asterisk, probably on the same box, as a protocol converter - you just need to tell it to use a non-standard port in sip.conf so that it doesn't clash with FreeSWITCH. --Dave
the lib that we used to provide iax support is pretty much abandonware (no longer updated) and newer iax implementations (like latest asterisk) can cause it to crash. There are no license compatible iax implementations that work, so.. mod_iax has been moved to the unsupported column.
Default passwords -- that is a single var in vars.xml that controls the passwords.
number ranges - up to you. The sample configs supplied are just that, samples. I use a smaller range personally.
On Fri, Feb 5, 2010 at 7:53 AM, Matthew Law <matt@webcontracts.co.uk> wrote: Why is that? - a lot of web pages I have read claim that IAX is more secure and efficient. I have no problem with using SIP whatsoever and it certainly appears to be ubiquitous. I am a complete newcomer to VoIP and I am trying to do this as securely as possible since I want to run freeswitch on a Xen VPS which will be visible on the internet.
Anyway, since my first question, I have worked my way through the wiki, read a lot of example configs and made some sense of the docs. I now have a very basic config working (with SIP) on a local vmware image using the 'quick and dirty' Makefile method. I removed all of the example configs from the xml files (those default extensions and passwords scared me) and added just one for extension 1000, plus my dialplan and inbound/outbound settings.
One question: is there any reason not to use a smaller extension number range, like 200-210, for example?
Now to figure out how to get time based roaming working...
Thanks,
Matt.
On Fri, February 5, 2010 6:43 am, Michael Jerris wrote: > iax2 support has been removed from FreeSWITCH in current trunk and will > not be in the 1.0.5 release. >
> Mike
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hi, iax2 is secure but, is not a good idea to avoid rtp and pass all packet including audio and signalisation troug the same port and digium added some change to the IAX2 protocol so freeswitch is not up to date no one want to update the iax2 stack in fs so fs mod_iax have bean removedfrom the trunk Le 05/02/2010 17:03, David Knell a écrit :
There's a fairly simple solution to IAX needs, which is to run Asterisk, probably on the same box, as a protocol converter - you just need to tell it to use a non-standard port in sip.conf so that it doesn't clash with FreeSWITCH.
--Dave
the lib that we used to provide iax support is pretty much abandonware (no longer updated) and newer iax implementations (like latest asterisk) can cause it to crash. There are no license compatible iax implementations that work, so.. mod_iax has been moved to the unsupported column.
Default passwords -- that is a single var in vars.xml that controls the passwords.
number ranges - up to you. The sample configs supplied are just that, samples. I use a smaller range personally.
On Fri, Feb 5, 2010 at 7:53 AM, Matthew Law<matt@webcontracts.co.uk> wrote: Why is that? - a lot of web pages I have read claim that IAX is more secure and efficient. I have no problem with using SIP whatsoever and it certainly appears to be ubiquitous. I am a complete newcomer to VoIP and I am trying to do this as securely as possible since I want to run freeswitch on a Xen VPS which will be visible on the internet.
Anyway, since my first question, I have worked my way through the wiki, read a lot of example configs and made some sense of the docs. I now have a very basic config working (with SIP) on a local vmware image using the 'quick and dirty' Makefile method. I removed all of the example configs from the xml files (those default extensions and passwords scared me) and added just one for extension 1000, plus my dialplan and inbound/outbound settings.
One question: is there any reason not to use a smaller extension number range, like 200-210, for example?
Now to figure out how to get time based roaming working...
Thanks,
Matt.
On Fri, February 5, 2010 6:43 am, Michael Jerris wrote: > iax2 support has been removed from FreeSWITCH in current trunk and will > not be in the 1.0.5 release. >
> Mike
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Its not that we didn't want to... nobody stepped up to help out so we had no choice but to tag it as unsupported. /b On Feb 5, 2010, at 3:15 PM, Meftah Tayeb wrote:
no one want to update the iax2 stack in fs so fs mod_iax have bean removedfrom the trunk
I had random crashes on IAX outgoing calls in mod_iax (all calls went to the same provider). I gave up and now use asterisk as protocol converter. On Friday 05 February 2010, Meftah Tayeb wrote:
hi, iax2 is secure but, is not a good idea to avoid rtp and pass all packet including audio and signalisation troug the same port and digium added some change to the IAX2 protocol so freeswitch is not up to date no one want to update the iax2 stack in fs so fs mod_iax have bean removedfrom the trunk
Le 05/02/2010 17:03, David Knell a écrit :
There's a fairly simple solution to IAX needs, which is to run Asterisk, probably on the same box, as a protocol converter - you just need to tell it to use a non-standard port in sip.conf so that it doesn't clash with FreeSWITCH.
--Dave
the lib that we used to provide iax support is pretty much abandonware (no longer updated) and newer iax implementations (like latest asterisk) can cause it to crash. There are no license compatible iax implementations that work, so.. mod_iax has been moved to the unsupported column.
Default passwords -- that is a single var in vars.xml that controls the passwords.
number ranges - up to you. The sample configs supplied are just that, samples. I use a smaller range personally.
On Fri, Feb 5, 2010 at 7:53 AM, Matthew Law<matt@webcontracts.co.uk> wrote: Why is that? - a lot of web pages I have read claim that IAX is more secure and efficient. I have no problem with using SIP whatsoever and it certainly appears to be ubiquitous. I am a complete newcomer to VoIP and I am trying to do this as securely as possible since I want to run freeswitch on a Xen VPS which will be visible on the internet.
Anyway, since my first question, I have worked my way through the wiki, read a lot of example configs and made some sense of the docs. I now have a very basic config working (with SIP) on a local vmware image using the 'quick and dirty' Makefile method. I removed all of the example configs from the xml files (those default extensions and passwords scared me) and added just one for extension 1000, plus my dialplan and inbound/outbound settings.
One question: is there any reason not to use a smaller extension number range, like 200-210, for example?
Now to figure out how to get time based roaming working...
Thanks,
Matt.
On Fri, February 5, 2010 6:43 am, Michael Jerris wrote: > iax2 support has been removed from FreeSWITCH in current
trunk and will
> not be in the 1.0.5 release. > > > > Mike
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exactly, The random crashes started happening by themselves, the protocol on the other end has exploited it's ancestor code and we really don't feel like bothering to fix it. On Fri, Feb 5, 2010 at 3:32 PM, Sergey Okhapkin <sos@sokhapkin.dyndns.org>wrote:
I had random crashes on IAX outgoing calls in mod_iax (all calls went to the same provider). I gave up and now use asterisk as protocol converter.
On Friday 05 February 2010, Meftah Tayeb wrote:
hi, iax2 is secure but, is not a good idea to avoid rtp and pass all packet including audio and signalisation troug the same port and digium added some change to the IAX2 protocol so freeswitch is not up to date no one want to update the iax2 stack in fs so fs mod_iax have bean removedfrom the trunk
Le 05/02/2010 17:03, David Knell a écrit :
There's a fairly simple solution to IAX needs, which is to run Asterisk, probably on the same box, as a protocol converter - you just need to tell it to use a non-standard port in sip.conf so that it doesn't clash with FreeSWITCH.
--Dave
the lib that we used to provide iax support is pretty much abandonware (no longer updated) and newer iax implementations (like latest asterisk) can cause it to crash. There are no license compatible iax implementations that work, so.. mod_iax has been moved to the unsupported column.
Default passwords -- that is a single var in vars.xml that controls the passwords.
number ranges - up to you. The sample configs supplied are just that, samples. I use a smaller range personally.
On Fri, Feb 5, 2010 at 7:53 AM, Matthew Law<matt@webcontracts.co.uk> wrote: Why is that? - a lot of web pages I have read claim that IAX is more secure and efficient. I have no problem with using SIP whatsoever and it certainly appears to be ubiquitous. I am a complete newcomer to VoIP and I am trying to do this as securely as possible since I want to run freeswitch on a Xen VPS which will be visible on the internet.
Anyway, since my first question, I have worked my way through the wiki, read a lot of example configs and made some sense of the docs. I now have a very basic config working (with SIP) on a local vmware image using the 'quick and dirty' Makefile method. I removed all of the example configs from the xml files (those default extensions and passwords scared me) and added just one for extension 1000, plus my dialplan and inbound/outbound settings.
One question: is there any reason not to use a smaller extension number range, like 200-210, for example?
Now to figure out how to get time based roaming working...
Thanks,
Matt.
On Fri, February 5, 2010 6:43 am, Michael Jerris wrote: > iax2 support has been removed from FreeSWITCH in current
trunk and will
> not be in the 1.0.5 release. > > > > Mike
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-- 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 <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
From the beginning Digium made a big mistake when released libiax as independent from chan_iax code. Protocol changes/bug fixes in chan_iax never (or maybe rarely) went back to libiax. This ended up in 2 different IAX protocol implementations. I wish Digium make chan_iax as a wrapper on top of supported and actively developed libiax... On Friday 05 February 2010, Anthony Minessale wrote:
exactly,
The random crashes started happening by themselves, the protocol on the other end has exploited it's ancestor code and we really don't feel like bothering to fix it.
On Fri, Feb 5, 2010 at 3:32 PM, Sergey Okhapkin <sos@sokhapkin.dyndns.org>wrote:
I had random crashes on IAX outgoing calls in mod_iax (all calls went to the same provider). I gave up and now use asterisk as protocol converter.
On Friday 05 February 2010, Meftah Tayeb wrote:
hi, iax2 is secure but, is not a good idea to avoid rtp and pass all packet including audio and signalisation troug the same port and digium added some change to the IAX2 protocol so freeswitch is not up to date no one want to update the iax2 stack in fs so fs mod_iax have bean removedfrom the trunk
Le 05/02/2010 17:03, David Knell a écrit :
There's a fairly simple solution to IAX needs, which is to run
Asterisk,
probably on the same box, as a protocol converter - you just need to tell it to use a non-standard port in sip.conf so that it doesn't clash with FreeSWITCH.
--Dave
the lib that we used to provide iax support is pretty much abandonware (no longer updated) and newer iax implementations (like latest asterisk) can cause it to crash. There are no license compatible iax implementations that work, so.. mod_iax has been moved to the unsupported column.
Default passwords -- that is a single var in vars.xml that controls the passwords.
number ranges - up to you. The sample configs supplied are just that, samples. I use a smaller range personally.
On Fri, Feb 5, 2010 at 7:53 AM, Matthew Law<matt@webcontracts.co.uk> wrote: Why is that? - a lot of web pages I have read claim that IAX is more secure and efficient. I have no problem with using SIP whatsoever and it certainly appears to be ubiquitous. I am a complete newcomer to VoIP and I am trying to do this as securely as possible since I want
to
run freeswitch on a Xen VPS which will be visible on the
internet.
Anyway, since my first question, I have worked my way through the wiki, read a lot of example configs and made some sense of the docs. I now have a very basic config working (with SIP) on a local vmware
image
using the 'quick and dirty' Makefile method. I removed all of the example configs from the xml files (those default extensions and passwords scared me) and added just one for extension 1000, plus my dialplan and inbound/outbound settings.
One question: is there any reason not to use a smaller extension number range, like 200-210, for example?
Now to figure out how to get time based roaming working...
Thanks,
Matt.
On Fri, February 5, 2010 6:43 am, Michael Jerris wrote: > iax2 support has been removed from FreeSWITCH in current
trunk and will
> not be in the 1.0.5 release. > > > > Mike
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I find the secure and efficiency claims on IAX to be pretty much a farce. IAX offers less overhead on rtp, but the same thing can be accomplished with rtp using packetization, as for security, I don't see any credible claim on that. IAX also forces the program to sort out a ton of audio for different users going to 1 socket, something that a network stack is quite good at when using different ports, but is a lot more work where we are getting the packets. As for default passwords and users, of course I wouldn't use those in production, those are for you to see how the pieces work together out of the box. I wouldn't however quickly scrap the entire default config, just read through them and think about what you need and do not. The extension ranges you use is totally at your discretion. Mike On Feb 5, 2010, at 8:53 AM, Matthew Law wrote:
Why is that? - a lot of web pages I have read claim that IAX is more secure and efficient. I have no problem with using SIP whatsoever and it certainly appears to be ubiquitous. I am a complete newcomer to VoIP and I am trying to do this as securely as possible since I want to run freeswitch on a Xen VPS which will be visible on the internet.
Anyway, since my first question, I have worked my way through the wiki, read a lot of example configs and made some sense of the docs. I now have a very basic config working (with SIP) on a local vmware image using the 'quick and dirty' Makefile method. I removed all of the example configs from the xml files (those default extensions and passwords scared me) and added just one for extension 1000, plus my dialplan and inbound/outbound settings.
One question: is there any reason not to use a smaller extension number range, like 200-210, for example?
Now to figure out how to get time based roaming working…
Hi, Can someone assist me with how one would achieve what Michael Jerris suggested: "IAX offers less overhead on rtp, but the same thing can be accomplished with rtp using packetization..." ...? I have done tons of reading on cRTP etc & I must be missing something as there is no obvious stock standard Linux way to achieve what IAX give you? Kind Regards, Michael On Fri, Feb 5, 2010 at 5:54 PM, Michael Jerris <mike@jerris.com> wrote:
I find the secure and efficiency claims on IAX to be pretty much a farce. IAX offers less overhead on rtp, but the same thing can be accomplished with rtp using packetization, as for security, I don't see any credible claim on that. IAX also forces the program to sort out a ton of audio for different users going to 1 socket, something that a network stack is quite good at when using different ports, but is a lot more work where we are getting the packets. As for default passwords and users, of course I wouldn't use those in production, those are for you to see how the pieces work together out of the box. I wouldn't however quickly scrap the entire default config, just read through them and think about what you need and do not. The extension ranges you use is totally at your discretion.
Mike
On Feb 5, 2010, at 8:53 AM, Matthew Law wrote:
Why is that? - a lot of web pages I have read claim that IAX is more secure and efficient. I have no problem with using SIP whatsoever and it certainly appears to be ubiquitous. I am a complete newcomer to VoIP and I am trying to do this as securely as possible since I want to run freeswitch on a Xen VPS which will be visible on the internet.
Anyway, since my first question, I have worked my way through the wiki, read a lot of example configs and made some sense of the docs. I now have a very basic config working (with SIP) on a local vmware image using the 'quick and dirty' Makefile method. I removed all of the example configs from the xml files (those default extensions and passwords scared me) and added just one for extension 1000, plus my dialplan and inbound/outbound settings.
One question: is there any reason not to use a smaller extension number range, like 200-210, for example?
Now to figure out how to get time based roaming working…
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Well what IAX gives us on FreeSWITCH is random segfaults because the protocol changed in a way that made the IAX stack library we were using a deathtrap. The authors of IAX decided to change the protocol so that it was not backwards compatible with their own previous work and due to lack of resources we decided that removing it was the best way to achieve stability. IAX offers some nat relief with the 1 port strategy at the expense of the stack implementation in user space. Frankly the authors of this protocol failed to supply a robust reference implementation of the protocol in a standalone license free package. There was one small library designed specifically for a standalone client (that's the one we used) and as I explained, in the end, it failed us. It then boils down to the business question: Do I want to chase after marginal savings on my configuration and packet sizes at the expense of much fewer calls and a serious risk of catastrophic failure caused by a poor implementation? SIP has its flaws as well, we don't have enough time to get into that, but on FreeSWITCH it can scale much higher than IAX even when it was working better (before they changed it). You said you are new to VoIP (which is surprising based on your email addr) so from your perspective what is it that you are looking for? rtp packetization refers to sending more audio at once to reduce the overhead of IP headers. in FreeSWITCH you can configure your codecs to PCMU@60i for instance to bundle 3 typical packets into 1 BTW, No matter what you use, you will have little luck on a virtual machine as your first experience. I recommend you get a real box (multi-core 64bit) to play with. On Wed, May 26, 2010 at 3:19 AM, Michael Toop < michaelt@voxcore.voxtelecom.co.za> wrote:
Hi,
Can someone assist me with how one would achieve what Michael Jerris suggested: "IAX offers less overhead on rtp, but the same thing can be accomplished with rtp using packetization..." ...?
I have done tons of reading on cRTP etc & I must be missing something as there is no obvious stock standard Linux way to achieve what IAX give you?
Kind Regards,
Michael
On Fri, Feb 5, 2010 at 5:54 PM, Michael Jerris <mike@jerris.com> wrote:
I find the secure and efficiency claims on IAX to be pretty much a farce. IAX offers less overhead on rtp, but the same thing can be accomplished with rtp using packetization, as for security, I don't see any credible claim on that. IAX also forces the program to sort out a ton of audio for different users going to 1 socket, something that a network stack is quite good at when using different ports, but is a lot more work where we are getting the packets. As for default passwords and users, of course I wouldn't use those in production, those are for you to see how the pieces work together out of the box. I wouldn't however quickly scrap the entire default config, just read through them and think about what you need and do not. The extension ranges you use is totally at your discretion.
Mike
On Feb 5, 2010, at 8:53 AM, Matthew Law wrote:
Why is that? - a lot of web pages I have read claim that IAX is more secure and efficient. I have no problem with using SIP whatsoever and it certainly appears to be ubiquitous. I am a complete newcomer to VoIP and I am trying to do this as securely as possible since I want to run freeswitch on a Xen VPS which will be visible on the internet.
Anyway, since my first question, I have worked my way through the wiki, read a lot of example configs and made some sense of the docs. I now have a very basic config working (with SIP) on a local vmware image using the 'quick and dirty' Makefile method. I removed all of the example configs from the xml files (those default extensions and passwords scared me) and added just one for extension 1000, plus my dialplan and inbound/outbound settings.
One question: is there any reason not to use a smaller extension number range, like 200-210, for example?
Now to figure out how to get time based roaming working…
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On Fri, 5 Feb 2010, Matthew Law wrote:
Why is that? - a lot of web pages I have read claim that IAX is more secure and efficient. I have no problem with using SIP whatsoever and it certainly appears to be ubiquitous. I am a complete newcomer to VoIP and I am trying to do this as securely as possible since I want to run freeswitch on a Xen VPS which will be visible on the internet.
Sigh. the web pages are wrong. I have implemented the earlier version of IAX2 inside the opal library. At one stage, I could make a voip call using IAX2 to digiums test server. The standard IAX2 codebase did not (at one stage) support silence suppression. With silence suppression, the bandwidth usage is halved. Using efficient packing of audio makes a few percent of difference. Doing silence suppression makes 50% of difference to bandwidth. IAX2 is more efficient? Most(>90%) of the cpu work when doing voip is in the codec. By using a non standard approach (iax2 audio packets) for carrying audio, you will have a minimal gain in effiency (or cpu usage). Remember that iax2 uses the same codecs as in H.323/SIP. Oh - you are right, g711 is a codec, and has trivial cpu cost. However, with g711 any bandwidth you save on the header is negligible compared to the size of the encoded audio block. Secure? Security protocols (HIP, ZRTP, etc) take years of careful development to get something that works well. I do not recall seeing evidence of years of development going into IAX2 security. Either the author of the security in IAX2 is a pure genius, able to do in days/weeks what other do in years,,,, or the iax2 security is average (at best). Yes, all the iax2 packets go to the same port. This has huge advantages for getting through firewalls, and setting up firewalls to accept incoming calls. It does (slightly) increase the complexity inside the iax2 code. =================================== if you want iax2 inside freeswitch, my suggestion is that the opal library needs to have the iax2 code there brought up to spec, and then used (in the same way as the H.323 component of the opal library). Derek. -- Derek Smithies Ph.D. IndraNet Technologies Ltd. Email: derek@indranet.co.nz ph +64 3 365 6485 Web: http://www.indranet-technologies.com/
Thanks for the comprehensive reply. I have switched to using plain SIP and the experiments continue... Matt On Sat, February 6, 2010 11:11 pm, Derek Smithies wrote:
Sigh. the web pages are wrong.
I have implemented the earlier version of IAX2 inside the opal library. At one stage, I could make a voip call using IAX2 to digiums test server.
The standard IAX2 codebase did not (at one stage) support silence suppression. With silence suppression, the bandwidth usage is halved. Using efficient packing of audio makes a few percent of difference. Doing silence suppression makes 50% of difference to bandwidth.
IAX2 is more efficient? Most(>90%) of the cpu work when doing voip is in the codec. By using a non standard approach (iax2 audio packets) for carrying audio, you will have a minimal gain in effiency (or cpu usage). Remember that iax2 uses the same codecs as in H.323/SIP. Oh - you are right, g711 is a codec, and has trivial cpu cost. However, with g711 any bandwidth you save on the header is negligible compared to the size of the encoded audio block.
Secure? Security protocols (HIP, ZRTP, etc) take years of careful development to get something that works well. I do not recall seeing evidence of years of development going into IAX2 security. Either the author of the security in IAX2 is a pure genius, able to do in days/weeks what other do in years,,,, or the iax2 security is average (at best).
Yes, all the iax2 packets go to the same port. This has huge advantages for getting through firewalls, and setting up firewalls to accept incoming calls. It does (slightly) increase the complexity inside the iax2 code.
===================================
if you want iax2 inside freeswitch, my suggestion is that the opal library needs to have the iax2 code there brought up to spec, and then used (in the same way as the H.323 component of the opal library).
participants (10)
-
Anthony Minessale
-
Brian West
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David Knell
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Derek Smithies
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Matthew Law
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Meftah Tayeb
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Michael Jerris
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Michael Toop
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Rupa Schomaker
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Sergey Okhapkin