one question,is it possible to change sip port on rpg 500?
From the interface,i can only enable and disable sip,but can i change sip port from 5060 to something else?
Maybe via cli?
Solved! Go to Solution.
no, it´s not possible. For reference have a look at the Admin Guide. Under "Port Usage" you can have a look at all needed ports and if they are configurable. SIP is not configurable...
i know this,i have an issue with RPG 500 behind a juniper firewall,while H323 is operational,there is no SIP incoming and outgoing at all.
I tried to reproduce this scenario but behind sophos UTM and i can see that outgoing is OK but i can not receive incoming,call is not establish,maybe sip signaling is mulfunctioning,that's why i would like to change this port.
Both firewalls are correctly configured,any ideas?
But the Group is successfully registered to SIP-Server/Registrar? What Software version is installed on Group? If you want to troubleshoot, start with Wireshark and check what SIP error message you receive.
i have traced the port,it seems that Polycom sends his internal IP to contact header,i think it should send its public,to be like 100.100.10.100:5060 but now it send 192.168.1.10:5060,so the other settings does not know where to respond.
With exactly the same firewall settings,i replaced Polycom with a video conference of another vendor,with the same internal ip,same public NAT and it worked.
Maybe it is a bug for 6.1.7-6.1.8 version,i will downgrade to test 6.1.5
i do not have a sip server/registrar, i just want to use SIP for a simple IP call,is it mandatory to register the device to a sip server?
Yes of course!! You always need an SIP-Server for communication! If you want direct communication, use H.323. I think you should read a bit about the topic VoIP, then it gets clearer for you to understand.
hello, the sip registrar server is just a "directory" and call signalisation manager no ?
we dont need it if we know the IP adresss of the sip terminal we want to join.
if you dial sip:10.10.10.10:5060 (directly the IP adress) you can call a sip terminal using sip. exactly likes h323 do)
i think his problem is due to NAT.
SIP and NAT can be hard to troubleshoot (depending of the device that performing NAT)
Many Brand (F5, checkpoint, cisco, Palo ..) of firewall/router/Loadbalancer device include help menchanism to help for thoses problems.
But even we these mechanism, you need to have a strong knowledge in the SIP protocol to resolve the problem if if doesnt work.
That's why many company uses public IP address directly for their visioconférences...
In my opinion this is no SIP, you make an IP call on this port, nothing else.
Two Groups, in the same IP subnet. How do they know what codec to use? How do they know what is their extension numbers? How do they know what ports to use? UDP? TCP, etc.?
In my opion they need SOMETHING to manage their configuration and push their configurations. This something is a call server (SIP-Server).