I have ASTERISK server with IP PUBLIC, and GSM VOIP GATEWAY behind NAT Dynamic. To setup trunking from GSM VOIP GATEWAY to Asterisk, i setup more PPTP VPN on asterisk server, so GSM VOIP GATEWAY can easily connect trunk with Asterisk over PPTP. Everything work very good, the only thing, sometime, RTP Audio lost packet and jitter.
I know this problem because I capture packet on asterisk server and monitor RTP stream on GSM VOIP GATEWAY.
I also increase MTU size on GMS VOIP GATEWAY to 1500 , but this problem less improvement. Please give some recommend.
Tks ycaner have mentioned my problem. Because my MODEM dont have IP public, But you think solution NAT better than PPTP , in my situatiuon? NAT will resolv lost packet and jitter?
I thinking , find a solution be optimize VPN Setting
Lost packets and jitter are ultimately the result of the network. PPTP appears to use a connectionless transport for the main data flow so it will allow packet loss and should not significantly worsen jitter (allowing packet loss is desirable, as the alternative is severe jitter). The one way it could make things worse is that it may hide Quality of Sesrvice markings, but it is extremely unlikely that your mobile data network would honour those anyway.
I would say your real problem is that you have just too many network components that are not suitable for isochronous data. The first thing you need to do is to replace the mobile network by a landline one, preferably one that is explicitly VoIP friendly. If the network is VoIP friendly, you should also remove the encapsulation, so that it can recognise the VoIP traffic.
I also notice that your diagram shows what look like wifi antennas. Going over wifi is also contra-indicated for good quality VoIP.
hello,
get a static IP from ISDN and you can get secure with static IP. You can use DMZ applicaton in Modem (Probable Gsdl) so there will be no NAt problem. Asterisk already handle Nat issue.
in addition Dinstar has a options “Onyl accepts call on ACL” . you can use it for secure.
Good luck.
