VOIP Vs VPN
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@Lost_Signal773 said:
Wrapping real time UDP based protocals with TCP is in general a waste of bandwidth (your going to retransmit data that will actually make the quality worse if it is processed out of order).
Yeah, not a good idea.
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@Lost_Signal773 said:
@NetworkNerd I'd like to point a few things. 1
- QoS over the internet means nothing. Your carrier is not going to respect your tags unless its on a P2P or MPLS circuit.
No, but QoS before it hits the Internet does the majority of what most people need. It is normally your own WAN link that is the choke point, not the open Internet. If it was, nothing we expect to work would work.
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@scottalanmiller said:
@Lost_Signal773 said:
@NetworkNerd I'd like to point a few things. 1
- QoS over the internet means nothing. Your carrier is not going to respect your tags unless its on a P2P or MPLS circuit.
No, but QoS before it hits the Internet does the majority of what most people need. It is normally your own WAN link that is the choke point, not the open Internet. If it was, nothing we expect to work would work.
Yep - that is how I should have worded it. Sorry if that was not clear from my post.
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Hi guys
i was unclear in my previous posts, what i have to say is :
when a sip client 192.168.1.10 want to talk with sip client 10.10.10.6 the following happen :
since we are connected via openvpn tunnel no NAT happen actually, because i run wireshark in both machine (remote vpnclient and IP PBX) and i saw that the packets carry the real IP of the sip phone not the public ip (source ip : 10.10.10.6 --- destination ip : 192.168.1.10)
what happened actually is routing not Nating, because there is a virtual tunnel between vpn client and vpn server and the vpn server route the packets to it destination (in this case 192.168.1.10 )
so the RTP traffic should be established without any issue since there is not NAT in the tunneli find the solution and now i have 2 site connected via openvpn and both location can call each other, the problem occur because i had wrong setting (in Ozeki server), before i set Discover public IP address using STUN server: stun.ekiga.net, this what created problem for me, i change it to my local IP PBX (Ozeki server) then everything work fine except sound quality, both side can hear each other but the sound was not totally clear (mixed with some noise),
any suggestion to improve sound quality knowing that in the main office the download speed is 4 mbs and upload speed is 0.5 mbs , and in the branch office we have download speed is 10 mbs and upload speed is 2 mbs ,
i think that my connection speed which decrease sound quality ????? or maybe something else ??? any suggestion ??thank you all for you answers
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That is a very tiny network connection. With only .5MB you will almost certainly have issues.
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What codec are you using!
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hiiiiiiiiiiiiiiiii Mr Scott, how are you doing
first of all, the available audio codec that i have in my PBX system is:
PCMA (8)
PCMU (0)
telephone-event (101)
iLBC (98)
G722 (9)
GSM (3)
SPEEX (97)
SPEEX (100)
SPEEX (108)
L16 (103)
L16 (11)
L16 (10)
G728 (15)
G723 (4)
G726-16 (104)
G726-24 (105)
G726-32 (106)
G726-40 (107)i select only 3 : PCMA (8) + PCMU (0) + GSM(3)
best regard
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i tried to select only GSM(3) since it is compatible with low bandwith but the call cannot be established, so i selected the additional 2 codec
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Try g.723 and make it preferred so that that is tried first.
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unfortunatlly i don't have this option of preference,
which are the other codec that i have to select with this codec, can i select it only or other additional codec has to be selected alongside ???thanks
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You can enforce a codec by removing all others.
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aaah you mean i select only one codec (g.723)
ok i got it
thanks -
Yes. That's not ideal but should work fine, at least for testing.
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I have had good luck with iLBC as well.
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great, is there any other thing that affect QOS ?? separating voice and data (VLANs) maybe a good idea (hhhhhh i know that you hate VLANs) i'm joking Mr scott
have a great day Sir
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QoS on the WAN links is definitely a big deal.
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but i have no control over the wan traffic, after the packets come out from the wan interface i can't control it !!!
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@IT-ADMIN said:
but i have no control over the wan traffic, after the packets come out from the wan interface i can't control it !!!
But you control them before they go out and that is 99% of the battle.
You can't control bullets after they leave a barrel but the order that you send them out pretty much guarantees the order in which they arrive.
The traffic that you have to worry about is your own and you control that.
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Yeah Sir you are right, i think that my firewall pfSense can do that because by way of it i can make queues, traffic priority, QOS, by way of these tools i think i can ensure that real time traffic have the highest priority than other type of traffic,
see you Mr Scott,,,,,,thank you
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@IT-ADMIN said:
Yeah Sir you are right, i think that my firewall pfSense can do that because by way of it i can make queues, traffic priority, QOS, by way of these tools i think i can ensure that real time traffic have the highest priority than other type of traffic,
see you Mr Scott,,,,,,thank you
Yes, pfSense can do that. @mike-ralston installed pfSense in the NTG Labs just yesterday too.