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SIP RTP

Question asked by aaron rinder on Jan 21, 2019
Latest reply on Jan 21, 2019 by Roman Niewiadomski

Hi all I have a customer who has a couple of unify sip phones for home users, they register via checkpoint firewall nat address to sip server this works ok, but on establishing a call there is no speech in either direction, after doing some traces I can see the rtp packets leaving the home users phone to the nat address and on the internal side I can see rtp packets being sent to the ip address of my phone from the sip server but no speech, customer has rules to allow udp ports from 5005- 65000 has anyone come across this before or has anyone got a bit more of a detailed configuration for sip phones connecting in this way

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