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
