Voip and RB750. that should be considered?
As we could solve the NAT, jiter, any recommendation?
Re: Voip and RB750. that should be considered?
If you're looking for some "magic" box which you plug in to solve all your VoIP problems, you'll be disappointed with everything. You need a deep and wide understanding of VoIP and Networking to be able to properly setup high quality and reliable VoIP.
That being said, your RB750 is not likely adding jitter or causing NAT problems if it's properly configured. Your provider is telling you that because either they don't know what their doing. They also likely told you to, "reboot" and then "reinstall".
That being said, your RB750 is not likely adding jitter or causing NAT problems if it's properly configured. Your provider is telling you that because either they don't know what their doing. They also likely told you to, "reboot" and then "reinstall".
Re: Voip and RB750. that should be considered?
Then they can reach an optimal configuration for VoIP and navigation to work for example in a network of 14 pcs and 2 VoIP phones 3 Mb ADSL downstream and upstream 400 K? . Or would be better to make a configuraación by Layer 7?
THx
THx
Re: Voip and RB750. that should be considered?
You need proper QOS, especially for the uplink (you can not do it downstream anyway, not under your control). Particularly as 400kb is TINY so if your VOIP packets do not get a higher priority queue, you pretty much are dead when someone does a download.
That is about it. NAT is SIP aware good enough, with a proper SIP server. Jitter is something that the phones are supposed to handle - naturally within reason, but outside that it is not Jitter but "broken network, establish QOS".
No problem.
Layer 7 is totally unneeded. I suggest using client side QOS to establish connection priority then work router side from the already set priority in the packets. Phones should allow you to set prioritization and for windows phones - windows can do it, QOS is hidden in the OS via group policy, google helps
That is about it. NAT is SIP aware good enough, with a proper SIP server. Jitter is something that the phones are supposed to handle - naturally within reason, but outside that it is not Jitter but "broken network, establish QOS".
No problem.
Layer 7 is totally unneeded. I suggest using client side QOS to establish connection priority then work router side from the already set priority in the packets. Phones should allow you to set prioritization and for windows phones - windows can do it, QOS is hidden in the OS via group policy, google helps
Re: Voip and RB750. that should be considered?
I comment on NAT, because 100 calls have no audio 8 on one side, from what I read, would come to be a subject of Symmetric NAT, what do I know my provider, to implement NAT rules in the nat or filter ?
As my VOIP provider uses large ranges of ports and ips, to implement the rule becomes non-functional. How to make the router work differently to symmetric nat. I also disabled the sip helper, because my provider works with STUN.
The QoS I have, is for marking packages and adrees list for voip (or would be too convenient to do so by packet size, ie up to 100k, for example?). But now bothering me this much the theme of NAT.
As my VOIP provider uses large ranges of ports and ips, to implement the rule becomes non-functional. How to make the router work differently to symmetric nat. I also disabled the sip helper, because my provider works with STUN.
The QoS I have, is for marking packages and adrees list for voip (or would be too convenient to do so by packet size, ie up to 100k, for example?). But now bothering me this much the theme of NAT.
Re: Voip and RB750. that should be considered?
Do you have STUN server configured on your phones? Some VoIP servers require that each phone behind the nat registers using a different local port. For example, local port of phone1 is 5060, for phone2 5061, and so on..I comment on NAT, because 100 calls have no audio 8 on one side, from what I read, would come to be a subject of Symmetric NAT, what do I know my provider, to implement NAT rules in the nat or filter ?
As my VOIP provider uses large ranges of ports and ips, to implement the rule becomes non-functional. How to make the router work differently to symmetric nat. I also disabled the sip helper, because my provider works with STUN.
The QoS I have, is for marking packages and adrees list for voip (or would be too convenient to do so by packet size, ie up to 100k, for example?). But now bothering me this much the theme of NAT.
Re: Voip and RB750. that should be considered?
Yes,Do you have STUN server configured on your phones? Some VoIP servers require that each phone behind the nat registers using a different local port. For example, local port of phone1 is 5060, for phone2 5061, and so on..I comment on NAT, because 100 calls have no audio 8 on one side, from what I read, would come to be a subject of Symmetric NAT, what do I know my provider, to implement NAT rules in the nat or filter ?
As my VOIP provider uses large ranges of ports and ips, to implement the rule becomes non-functional. How to make the router work differently to symmetric nat. I also disabled the sip helper, because my provider works with STUN.
The QoS I have, is for marking packages and adrees list for voip (or would be too convenient to do so by packet size, ie up to 100k, for example?). But now bothering me this much the theme of NAT.
on the phone, this set the stun server, and also each phone has a different port, as you mention.
It is difficult to determine which ports in sip SIP TCP Port Min says: 5060; SIP TCP Port Max: 5080.
RTP Port Min: 20384 RTP Port Max 20482.
RTP Packet Size: 0.020.
Re: Voip and RB750. that should be considered?
Code: Select all
/ip firewall mangle
add action=mark-packet chain=postrouting disabled=no dst-address-list=VoIP new-packet-mark=voip_out out-interface=ether1 passthrough=no protocol=udp
add action=mark-packet chain=prerouting disabled=no in-interface=ether1 new-packet-mark=voip_in passthrough=no protocol=udp src-address-list=VoIP
add action=mark-packet chain=postrouting disabled=no new-packet-mark=voip_out out-interface=ether1 packet-size=0-100 passthrough=no protocol=udp
add action=mark-packet chain=prerouting disabled=no in-interface=ether1 new-packet-mark=voip_in packet-size=0-100 passthrough=no protocol=udp
add action=mark-packet chain=postrouting disabled=no new-packet-mark=internet_out out-interface=ether1 passthrough=no
add action=mark-packet chain=prerouting disabled=no in-interface=ether1 new-packet-mark=internet_in passthrough=no
Code: Select all
/queue tree
add burst-limit=0 burst-threshold=0 burst-time=0s disabled=no limit-at=0 max-limit=2500k name=TotalIN packet-mark=ALL-IN parent=global-in priority=1
add burst-limit=0 burst-threshold=0 burst-time=0s disabled=no limit-at=0 max-limit=350k name=TotalOut packet-mark=ALL-OUT parent=global-out priority=1
/queue tree
add burst-limit=0 burst-threshold=0 burst-time=0s disabled=no limit-at=300k max-limit=500k name=VOIP_IN packet-mark=voip_in parent=TotalIN priority=1 queue=pcq-down
add burst-limit=0 burst-threshold=0 burst-time=0s disabled=no limit-at=150k max-limit=200k name=VOIP_OUT packet-mark=voip_out parent=TotalOut priority=1 queue=pcq-up
add burst-limit=0 burst-threshold=0 burst-time=0s disabled=no limit-at=0 max-limit=2200k name=INTERNET_IN packet-mark=internet_in parent=TotalIN priority=2 queue=pcq-down
add burst-limit=0 burst-threshold=0 burst-time=0s disabled=no limit-at=0 max-limit=200k name=INTERNET_OUT packet-mark=internet_out parent=TotalOut priority=2 queue=pcq-up
/routing bgp instance