Some significant problems arise when using compression codecs. G.711 provides toll quality and has much better error recovery.
Silence suppression can also impact the voice quality since there is always a transition between silense and talking.
Once test I would like to be able to perform is multiple codec processing of the same audio sample. For example, take a standard voice sample, process is through a G.711 codec, Conver it back to anolog through anoother G.711 processer then process it again through a G.729a codec. Having some PSQM scores on this would be cool. This is what will happen going from a VoIP network to the PSTN back onto a VoIP network using two different codecs. Any suggestions?
Sequention packet loss significantly affects voice quality with compression codecs. QoS is more than bandwidth shaping.
Bandwidth shaping attempts to limit the influence of other protocols on a voice converation (in reagrds to VoIP).
RSVP and other protocols are intended to ensure adequate bandwidth is avaialble before allowing the call to start. Other mechanisms exist such as a static number of calls alows per link and is managed by the VoIP system its self.
Pardon the typos, this little box is hard to type into.
Ramblings from Webpro…