Tue May 26, 2009 1:51 am
Assuming you are converting to VoIP of some type, yes it can be done. By this I mean that there is no built-in E1 TDM transport like that available on some PTP radios. Unless you are using NStream2, you will have a half-duplex link. This means you need to have good enough throughput to be able to transport the 2400Kbps up and 2400Kbps down at all times to support a full 30 channels of aLaw or uLaw RTP streams. This is completely possible with a pair of RB411AH boards and applicable radio cards and antennas. If you want the PTP link to also handle data as well, you have to use the Queue Trees and Mangles to ensure you are appropriately bandwidth shaping and providing QOS (i.e. the RTP always gets 2400Kbps and always goes first).
People get themselves into trouble trying to do this usually make one or more of these mistakes:
1) Cheap and/or undersized antennas for the shot distance
2) Poor Line-Of-Sight (LOS), usually caused by a lot of freznel blockage
3) Attempting to get too much throughput, e.g. they think that the 54Mbps modulation scheme allows them to sometimes, or always, get 54Mbps of IP throughput. If you get 30Mbps half-duplex, you're doing great. In the VoIP case, you need enough throughput to reliable emulate a full-duplex link at lower speeds. In other words, if you can get 25Mbps of UDP throughput in one direction (half-duplex), then you can probably get a reliable 10Mbps full-duplex.
4) Not taking into account the half-duplex nature of the devices
5) Misunderstanding the difference between the control channel and the audio stream with VoIP. For example, with SIP, SIP is the call control and RTP over UDP is the audio stream transport scheme.
6) Lack of full understanding of NAT and it's interaction with VoIP protocols
If you are converting from E1 to IP and then back to E1, you have to have some equipment which will reliably recreate E1 clocking on the other side. Adtran and RAD both make boxes to do this.