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Re: hosted PBX/VOIP thru VPN?
On Nov 11, 2008, at 6:17 PM, Lorell Hathcock wrote:
The implementation of a VPN indeed would probably not result in an improvement of your customer' RTP packet delivery, either for jitter or packet loss. If you wish to see if RTP is being meddled with, try changing the RTP port numbers on the ATA or on the remote side to something less typical of the RTP port range - try something <10000. While some deep-packet inspections will dig through each packet to categorize and stomp on RTP voice audio, it is probably not the case that anyone in the path you describe is applying anything other than port number and protocol (UDP/TCP) inspections, if they are doing any such punitive QOS at all.
I would be very interested to learn if you or anyone does know of a transit carrier who is de-pref'ing RTP packets as a peered transit provider (or non-paid peering partner.) This doesn't mean static "end customers" - I'm really talking about traffic that is ingress/egress from some other ASN, even if that ASN is paying for transit. This would be a fairly major departure from any type of QOS or de- preferencing that I've seen before, and I'm sure the list would be interested in any results as well.
The root cause of the problem your customer describes also needs to be identified - that will tell you a lot. Wireshark a few calls and see what you can see on the RTP packet path. Without more specifics on the "bad performance", it will be difficult to determine if this is even a network issue at all - maybe it's just a sub-standard ITSP, gateway, or even PSTN path on the far side of the equation.
A very slight chance exists that due to round-robin routing you are getting out-of-order packets in one or both directions of the media stream. RTP does not recover well from OOO packets. Try some traceroutes in the RTP port range to see what happens. You can see one direction for the traceroute UDP outbound and watch for multi- pathing, and you can see the other direction with wireshark on inbound OOO RTP streams to your customer. If the problem is out-of-order RTP packets, then there are some things that a GRE tunnel plus some creative route announcements/static routes might be able to solve, and those are left as an exercise for the reader. But a "VPN" is almost always going to be the wrong answer for VoIP performance increases - GRE is better suited for encapsulating UDP, and I run VoIP connections over GRE all the time due to the perverse and unfortunate routing situation for my home network, which resides at the end of a consumer- grade cable IP connection. I would not recommend even GRE as a matter of course for VoIP RTP transport; I merely say that it is possible, and in some fringe cases the only solution available.
FWIW: Snom (a SIP-based desk phone) now includes a built-in IPSEC tunnel protocol stack so the phone can securely establish connections back to the PBX or other endpoints. The reasons for doing this are not clearly not performance-oriented - they are security-oriented. It even will encapsulate traffic from any hosts attached to the one-port switch on the back. Desk phones are getting pretty scary. I'm waiting for "sh ip bgp" for my Cisco 7960...