jroper
Guru
- Joined
- Oct 20, 2007
- Messages
- 3,832
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- 71
Hi
I've finally taken the time to have a look at OpenSBC, which provides a number of functions, including RTP Proxy, upper registration, and Session Border controller.
For those who are doing commercial PBX installations, and regularly run into the issues of NAT, and one way audio, particularly where both ends are behind NAT, OpenSBC could be the answer.
Ideally OpenSBC should be placed on an external IP address for best results, but this is not necessarily required, from what I have read.
1. OpenSBC will do Upper Registration, that is to say that you point your phone at the OpenSBC server, and it will forward your registration onto the asterisk server. This means that those issues of delivering the right IP to the PBX go away. As far as asterisk is concerned, the IP address of the phone is the IP of the SBC.
2. OpenSBC will proxy the media - doing NAT traversal when only one end is behind NAT is an order of magnitude easier than doing NAT traversal when both ends are behind NAT.
3. One SBC can handle registrations and proxying RTP to multiple Asterisk servers by means of simple rules, but some forethought has to be given to planning this.
4. Greater security. Registrations can be set to only come from the SBC IP address on your IP-PBX, although some attention has to be given to security on OpenSBC, it does not run so many services as the average PBX, and should therefore be easier to lock down.
The interface is nasty, but functional, and once set up, you don't have to touch it again until you need to add another PBX to the mix.
Whether it is a silver bullet remains to be seen, but we should know after some testing, my testing today allows me to register two SIP phones behind the same NAT device, and allow two way audio. That was not possible before.
To those professionals out there who need a solution to one way audio, and NAT/SIP problems, if there is an interest in documenting the build of this, and some default configs, reply to this thread.
Yours
Joe
I've finally taken the time to have a look at OpenSBC, which provides a number of functions, including RTP Proxy, upper registration, and Session Border controller.
For those who are doing commercial PBX installations, and regularly run into the issues of NAT, and one way audio, particularly where both ends are behind NAT, OpenSBC could be the answer.
Ideally OpenSBC should be placed on an external IP address for best results, but this is not necessarily required, from what I have read.
1. OpenSBC will do Upper Registration, that is to say that you point your phone at the OpenSBC server, and it will forward your registration onto the asterisk server. This means that those issues of delivering the right IP to the PBX go away. As far as asterisk is concerned, the IP address of the phone is the IP of the SBC.
2. OpenSBC will proxy the media - doing NAT traversal when only one end is behind NAT is an order of magnitude easier than doing NAT traversal when both ends are behind NAT.
3. One SBC can handle registrations and proxying RTP to multiple Asterisk servers by means of simple rules, but some forethought has to be given to planning this.
4. Greater security. Registrations can be set to only come from the SBC IP address on your IP-PBX, although some attention has to be given to security on OpenSBC, it does not run so many services as the average PBX, and should therefore be easier to lock down.
The interface is nasty, but functional, and once set up, you don't have to touch it again until you need to add another PBX to the mix.
Whether it is a silver bullet remains to be seen, but we should know after some testing, my testing today allows me to register two SIP phones behind the same NAT device, and allow two way audio. That was not possible before.
To those professionals out there who need a solution to one way audio, and NAT/SIP problems, if there is an interest in documenting the build of this, and some default configs, reply to this thread.
Yours
Joe