SIP RTP

[aaron_rinder]

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

[Roman_Niewiado1 · Accepted solution]

Hi Aaron, it could be, that you need ProxyARP entry on the Firewall for the NAT IP.

[m_bilal]

I also face same issue R80.40 T180 some RTP connections not working. no zdebug drops, if we nat Object Static level we don't need to add manual proxy arp, CP will automatically get the arp configuration right? i am wrong please correct me.

Thank you.

[Chris_Atkinson]

Perhaps sk177365 will help you?