hi all ,
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.