Praise for the Beta Version 6

I have loaded Beta version 6 and have tested with VOIP. I have to say I think that this is the best I have ever seen Mikrotik perform. I have SIP VOIP running and wireless with QOS and it performs like it has never done before. Wireless seemd to work better and is more responsive. The only thing I can find wrong so far is the export and import. With the SIP NAT helper this works great.

PRAISE PRAISE PRAISE

Keep up the hard work

while I’m definitly happy with the progess I’ve seen so far on Beta 6 (I have a test link running pretty stable in an enviroment that 2.9.38 would not even link up at) it’s still a way from being ready for production. I’ve seen issues with the periodic calibration (already discuessed with support, disableing it seems to fix my issue)…

I’m curious to what QOS you setup for the VoIP traffic.. did you just setup a simple queue? we don’t use NAT anywhere so the SIP helper is of no use to us (not that it does anything for QOS)

How is Nstreme performing?

/Henrik

I have Mangle rules and Queue tree setup

What I have test inhouse with nstream seems to work. I can’t test with any distance.

in our initial testing, we saw a performance boosts of 20% or more in area that had lots of other 802.11 traffic. we’ve also seen increased resistance to noise that was previously preventing links from even being estalished, and now we’re able to push 1-2mbps across them, however ping times are not what I wuld like to see…

I can see a better signal value also. As far as ping times, This could be how the network is built. If you route it, ping time will be a little more than if you bridge. Both has advantages and disadvantages. Mikrotik seems to have higher ping times than staros. I have used both in building the network but I like Mikrotik moreover.

I have seen good things on mailfunctioning links, but with Nstreme enabled i get nothing but problems when loading it with traffic, this is both in lab and real setups, single/dual Nstreme setups.. A lot of link flapping on 532A when pushing traffic through..

/Henrik

gwinton,

Can you elaborate on how you’ve configured the SIP in Beta?

  1. I see Beta you can specify under /ip firewall service ports sip a port number. The default port number is blank. Did you have to put something in there?

  2. Which type of SIP configuration do you have:

   1.  Asterisk as a SIP client behind nat, connecting to outside SIP Proxies
   2. Asterisk as a SIP client behind nat, connecting to inside SIP proxies
   3. Asterisk as a SIP server behind nat, clients on the outside connecting to Asterisk
   4. Asterisk as a SIP server behind nat, clients on the outside behind a second NAT connecting to Asterisk
   5. Asterisk as a SIP server behind nat, clients on the inside connecting to Asterisk
   6. Asterisk as a SIP client outside nat, connecting to outside SIP proxies
   7. Asterisk as a SIP client outside nat, connecting to inside SIP proxies
   8. Asterisk as a SIP server outside nat, clients on the outside connecting to Asterisk
   9. Asterisk as a SIP server outside nat, clients on the inside connecting to Asterisk

I tried #4 with a difference: Dual NAT on client’s side, no joy.

  1. Did you use QoS with SIP? If so How?