Media delay / re-invite issue with unsupported VoIP provider

[mit]

It is not that the media stream passes through the 3CX server twice before reaching the PSTN, it is simply that one set of delay is incurred on the media stream from the A party to the B party, and then further delay is incurred on the return media stream from the B party to the A party. This means that the delay between the A party finishing speaking and then hearing the B party's response is the combination of both these delays, which is the round trip media delay.

An exmaple call flow is as follows:
A Party > IP Handset > 3CX Server > SBC > VoIP Provider > PSTN Gateway > PSTN > B Party
return speech path follows the same route in reverse, albeit via a different SBC

 

[pjr]

In our case it is certainly noticeable. Firstly the 100ms delay (in reality, in our case I think it is a bit longer than this) is incurred on each pass through the 3CX server, so on a conversation this adds a total additional delay of 200ms between parties speaking. Obviously any latency outside our network is additional and any jitter buffers at either end can add this, so in a typical example calling a number from the UK to the UK we might experience a media round trip delay of about 300ms.

This probably makes what would normally sound like a local call in the UK have an equivalent delay to calling between the UK and West Coast US or thereabouts (and when calling from the UK to the US it is accordingly worse). In our situation, having moved all users over one weekend from traditional TDM based PBX to a fully IP solution we have had comments from almost all users mainly related to awkwardness in conversation, where one party inadvertantly talks over the other.

I think the ITU reckons that the maximum acceptible end to end delay is 150ms, so on a call from the phone on my desk to another number in the same city my delay can be approaching this limit.

What I guess is quite frustrating is that this delay is not necessary, there is no need for me to transocde this audio stream on the 3CX server, it can remain as IP until it hits the PSTN, where the transcoding can be performed by a device capable of doing so without significant delay.

Hi Mit,

We have installed a few 3CX systems and use an unsupported VoIP provider (Gamma Telecom). We seem to have no problems with calls made from phones local to the PBX (ie phones on the same LAN as the PBX), but are experiencing intermittent delays on calls made by phones that are connected to the PBX via VPN. This doesnt occur on every call - but happens more often than not. Was the delay you experienced consistent on every call?

The phones that are connected via VPN have their own dedicated ADSL connections which show 20ms responses from the remote phone LAN to the 3CX server LAN.

What overhead did you find that the the internal media server adds to all calls made via SIP Providers?

Thanks

Paul

 

[mit]

Hi Paul,

Our problem is a consistent delay on every call. We do not use VPN in our setup so it is not a direct comparison. We found that when the calls traversed the media server it added probably 100-150ms of consistent delay to the calls. I guess you should be able to tell from the media server logs whether the media server is being invoked when you have your calls coming through on VPN.

Incidently, if you are using your VoIP provider for a call from a phone local to the PABX, do you know if you call is using the media server or if the media is direct in that situation (i.e. are reinvites working correctly).

One other item I picked up while investigating this issue is that somewhere in the advanced settings on the PABX you can set what networks are considered local. If the media server is being involved in the calls from you VPN connected phones, it might be worth taking a look at this setting.