[Asterisk-Users] Open ports?

2005-03-15 Thread Jacob Cazzell
Hi all, I have a quick question I hoping someone can help me with. I have [EMAIL PROTECTED] running and working just fine. I've integrated it with BroadVoice and so far I'm blown away by everything I can do. I don't particularly like sitting my entire machine in the DMZ on my network sitting

Re: [Asterisk-Users] Open ports?

2005-03-15 Thread Rich Adamson
I have a quick question I hoping someone can help me with. I have [EMAIL PROTECTED] running and working just fine. I've integrated it with BroadVoice and so far I'm blown away by everything I can do. I don't particularly like sitting my entire machine in the DMZ on my network sitting

[Asterisk-Users] Open Ports

2004-12-18 Thread Norman Zhang
Hi, May I ask what ports are necessary for SIP communication through a firewall? I read somewhere that UDP/5060 alone is enough. Some recommends more ports to be opened for RTP. Regards, Norman Zhang ___ Asterisk-Users mailing list [EMAIL PROTECTED]

Re: [Asterisk-Users] Open Ports

2004-12-18 Thread Antony Stone
On Saturday 18 December 2004 10:17, Norman Zhang wrote: Hi, May I ask what ports are necessary for SIP communication through a firewall? I read somewhere that UDP/5060 alone is enough. Some recommends more ports to be opened for RTP. Both the above statements are correct. SIP uses port

Re: [Asterisk-Users] Open Ports

2004-12-18 Thread Norman Zhang
May I ask what ports are necessary for SIP communication through a firewall? I read somewhere that UDP/5060 alone is enough. Some recommends more ports to be opened for RTP. Both the above statements are correct. SIP uses port 5060 RTP uses multiple ports, typically in the range 1-2

Re: [Asterisk-Users] Open Ports

2004-12-18 Thread Antony Stone
On Saturday 18 December 2004 10:58, Norman Zhang wrote: SIP uses port 5060 RTP uses multiple ports, typically in the range 1-2 Remember that SIP and RTP are different - SIP is used to set up the call; RTP is used to carry the audio once the call has been set up. Thanks. May

Re: [Asterisk-Users] Open Ports

2004-12-18 Thread Rich Adamson
May I ask what ports are necessary for SIP communication through a firewall? I read somewhere that UDP/5060 alone is enough. Some recommends more ports to be opened for RTP. Both the above statements are correct. SIP uses port 5060 RTP uses multiple ports, typically in the range

Re: [Asterisk-Users] Open Ports

2004-12-18 Thread Rich Adamson
SIP uses port 5060 RTP uses multiple ports, typically in the range 1-2 Remember that SIP and RTP are different - SIP is used to set up the call; RTP is used to carry the audio once the call has been set up. Thanks. May I ask what security control can be applied to RTP

Re: [Asterisk-Users] Open Ports

2004-12-18 Thread Antony Stone
On Saturday 18 December 2004 11:40, Rich Adamson wrote: But, to return to my initial question, what's the security risk in leaving your Asterisk server open to UDP packets from the world? I regard it like a mail server - a firewall allowing TCP packets through to port 25 cannot protect

Re: [Asterisk-Users] Open Ports

2004-12-18 Thread tim panton
Comments inline... On 18 Dec 2004, at 11:40, Rich Adamson wrote: SIP uses port 5060 RTP uses multiple ports, typically in the range 1-2 Remember that SIP and RTP are different - SIP is used to set up the call; RTP is used to carry the audio once the call has been set up. Thanks. May I ask

Re: [Asterisk-Users] Open Ports

2004-12-18 Thread Norman Zhang
Cisco phones use udp ports 16384-32776, while Xlite uses something like udp ports 8000-8050, and Polycom phones use another range, etc. If you worked for a large company that didn't have any sip phone standards and you had to open everything that _could_ be used for rtp, then you really would be

Re: [Asterisk-Users] Open Ports

2004-12-18 Thread Rich Adamson
Cisco phones use udp ports 16384-32776, while Xlite uses something like udp ports 8000-8050, and Polycom phones use another range, etc. If you worked for a large company that didn't have any sip phone standards and you had to open everything that _could_ be used for rtp, then you really