HOSTED SOLUTION AND RTP

[jholland]

How do you prevent all the RTP from being anchored by a hosted version of pbxnsip?

[Vodia PBX]

Well, realisticly is you want to provide hosted services you need a session border controller anyway. And that realisticly means that you need to relay the media anyway. And realisticly, statistically speaking, most of the calls are external calls anyway and there media path optimization does not give you anything.

 

The biggest problem is delay, and here especially DSL. If you have fast DSL or other access methods, the delay becomes less critical and then it does not really matter anymore if you relay or not.

 

And the good news is that you can easily offer features like call barge in and recording.

[jholland]

We are using a Session Border Controller and there is no need for the media to be anchored in both places. We are seeing underuns on the PBX so RTP is flowing through it so how do we bypass that?

[hosted]

We are using a Session Border Controller and there is no need for the media to be anchored in both places. We are seeing underuns on the PBX so RTP is flowing through it so how do we bypass that?

 

What SBC are you using with pbxnsip?

[Krom]

I am just joining the PBXnSIP community and have the same question as jholland posted back in March.

 

In our hosted environment we are seeing that about 40% of our calls are intercom calls or supervised transfers and would like to keep the local LAN (intercom) calls RTP media local with RE-INVITEs after the PBXnSIP has handled the call signaling to the registered end point. This would save of the bandwidth requirements between the hosted site and the remote location. It should also be the method for any group paging feature.

 

We are using OpenSER as the front end proxy between the remote location end points and PBXnSIP. The end points of course are registering with PBXnSIP and not OpenSER so I would be looking for a solution within PBXnSIP to support RE-INVITEs between end points registering from the same remote locations.

 

Again, I am new to PBXnSIP so please forgive any ignorance. But I have abandoned all hope for * in a hosted environment. SIP stack blows....

 

Krom

[Vodia PBX]

I heared that practically only ACME and Nextone are left in the SBC arena. They are hardware based. As far as I can tell they are both pretty good. Don't expect many blow's and whistles, but these devices are rock solid and that is what you expect from a SBC.

 

Interestingly, the do also RTP relay. On other words, once that the RTP makes it to the SBC, it is only a ms away from the PBX which makes the whole RTP relay discussion quite pointless. Jasomi tried to do "media path optimization", but IMHO in the real life the environments were so difficult that this did not work in all cases.

 

Maybe we will find out that the good old class4/class5 architecture was not so bad. With the difference that this time, the Centrex is an economical interesting alternative to the CPE-based IP-PBX.