VoIP from IP phone to PBX thought a OpenVPN tunnel

kaneda

Hello! Y have setup about ten OpenVPN tunels for some roadwarriors and remote clients.
Initially they only use a RDP connection to a Windows server but now we are giving them voIP phones.
Our config is an static IP on the client tun interfaz, udp protocol and an static route on the pfsense that guide the packets from pbx to the remote telephone subnet thought tap interface IP (windows routing enabled on clients.)

The telephone rings, you can talk, but the quality is poor, and when you speak fast or a lot of time (5-10 mins), the words sounds together like if it where congested or collapsed.

I have read from other people that dont have any problems with this conf but this is common for all my clients, only one at Germany with ADSL have better quality but cant speak very fast.

All others uses 3´5G modems, with HDSPA services, some of them have a fair upload/download (5Mb down/1Up)rates but appear to be not enought for voip.

PBX IP (ALCATEL OXO)- Pfsense - (OpenVPN) - Vodafone 3,5G net - Windows Client - Ip telephone.

I forgot to tell that Im using G711 Codec so no quality loss must appear.

Any Idea about how to improve the Voip links o whole OpenVpn Link to have better services?

Many thanks

eazydor

are you running openvpn over tcp packets?
also, try enabling real time data compression..

udp is only responsible for addressing, without securing the transmission of data, as this would lead to delays in speech.
tcp in contrast is there for transmit data controlled and reliable way, which will cause "delay" in voice or video, i.e…

compression expresses itself.

proto tcp-client –> proto udp
#comp-lzo --> comp-lzo

http://forum.pfsense.org/index.php/topic,22414.0.html

Acecare

@eazydor:

are you running openvpn over tcp packets?
also, try enabling real time data compression..

udp is only responsible for addressing, without securing the transmission of data, as this would lead to delays in speech.
tcp in contrast is there for transmit data controlled and reliable way, which will cause "delay" in voice or video, i.e…

compression expresses itself.

proto tcp-client –> proto udp
#comp-lzo --> comp-lzo

I definitely agree with you and I think he must change his current VoIP Phone if the possible solution doesn't work. I am using a Hosted VoIP Phone Systems and I have no problem of using it. It because they provide enough services to their customer and in terms of updating, you don't need to pay for it.

kaneda

Hello, thanks for replies, Im using udp, and tried with and without compression, and we did not improve the quality, our main problem appears to be latency, our system, alcatel OXO appear to need latency below 100ms, we need to mark whole tunnel as VoIP, and reserve bandwidth for it to lower the delay, any idea?