Message from Link2VoIP, my voip provider. I thought iax2 was more stable than sip. If so what drugs are they on?
Dear Link2VoIP Customers,
This is an e-mail alert advising all customers currently using the IAX2
(Inter-Asterisk eXchange) protocol for Asterisk and TrixBox that they must
switch over to SIP (Session Initiation Protocol) immediately as IAX2 will no
longer be supported as of February 28, 2007.
Why the change? We have found, through rigorous tests of our network
infrastructure that IAX2 as a solution simply does not have the full
stability required to handle thousands or more simultaneous calls required
from VoIP carriers such as ours.
Phasing out IAX in favor of SIP (for the purpose of enhancing the
reliability and stability) will allow our company to grow with the network.
As stated on the Link2VoIP News Section for January 11th, we are in the
process of changing and upgrading our equipment which will allow us to
handle more calls efficiently while maintaining the highest quality of
service for our customers - using SIP allows us to achieve this goal.
Customers who are unsure of how to switch from IAX2 over to SIP, examples
have been posted to the help page (after you select SIP under “settings”),
and for those companies who require additional technical assistance, may
e-mail the Link2VoIP Support Team: support@link2voip.com. Please include
your full name, username, and a full explanation of the issue you’re having
with the switch over, and we’ll assist you as soon as possible.
I think they got to big or are uninterested in supporting asterisk if they can run a switch that makes life a lot easier for them. IMHO I think this is a mistake since IAX has a lot of qualities that you will never find in SIP.
The quality of each does depend on the service provider that you are using. I found with Gradwell in the UK that using a SIP trunk provides better voice quality that IAX2. In their case SIP is provided directly from their CISCO Call Manager hardware whereas IAX2 simply provides another hop that then goes into the Call Managers.
Vitelity is also dropping on IAX in favor of SIP, and they say IAX has a bug. As we use both Vitelity and Link2Voip, we already switched to SIP for inbound and outbound and everything’s going fine.
from Vitelity
to Customer
date Jan 9, 2007 7:09 PM
subject Vitelity - Important Settings Change
Dear Customer,
Vitelity is now migrating all calls to our new network. This notice affects all users, including those put on special servers at our other locations.
You should now change Asterisk inbound trunks, registration strings and SIP ATA settings to use the host inbound2.vitelity.net in order to receive inbound calls. Outbound calls for softswitch and PBX customers should still go to outbound1.vitelity.net. If you require assistance, new settings are now available on the support page within the user portal found at portal.vitelity.net.
Asterisk-IAX users are strongly advised to switch to SIP as you will experience better sound quality. This is due to a bug in Asterisk that causes jitter during peak hours on IAX channels.
We thank you for your patience during this migration and look forward to serving you in the future.
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Please reply to this email if you require anything regarding the content of this email.
large providers have found it’s cheaper to run sip simply because it spans port usage over a wide array, limiting bandwidth overload. iax and iax2 run on a single port…
I hope everyone running Asterisk systems will drop their service provider if they no longer provide IAX2 as an option. Obviously the only thing that matters is money, so show them what you prefer and stop giving them yours. Switch to a provider that still provides IAX2 as an option.
[quote=“ericbechtel”]Hey guys…I know why this is…it’s not a bug
large providers have found it’s cheaper to run sip simply because it spans port usage over a wide array, limiting bandwidth overload. iax and iax2 run on a single port…[/quote]
when a single asterisk server has anything more then 50 concurrent IAX channels open, it becomes unstable and starts causing quality issues. choppyness, etc. with SIP, you can easily have over 90 channels before you will start to experience any quality problems. (this is from personal experience)
this is why larger asterisk providers are moving away from IAX.
There are to many variables involved to make a blanket statement like that. If I’m running a dual processor dual core 3Ghz machine with 4 gigs of ram directly trunked to an OC3, am I going to have that same limit?
Or, if I’m running a celeron 1Ghz Celeron with 256MB of ram on a 144k isdn, will I get 50 trunks?
In your example, are you only using one server? Because I doubt providers are only using one server.
core 2 duo 2.66ghz, 4gm ram, 10k rpm sata hd. of course we use multiple servers… but that doesn’t stop the fact that after a single server reaches 50 concurrent calls, the quality severely degrades. snap, crackle, pop.
Hmm… Maybe if I gets some time I will trunk a couple of Asterisk systems together on my gigabit network and see if I get the same results after opening 50 concurrent calls. Would hold music be good enough to simulate an active call, or should I loop a voice track through them?
at the time we did the testing, after approx 50 calls/channels opened the quality became horrible for iax users. switching to sip resolved the issues entirely.