It looks like there is no way to set up working QoS for VOIP phone, at least not using Linksys PAP2T adapter.
Correct me if I am wrong, but SIP travels over UDP ports specified per line, i.e. 5060 and 5061 in my case. I can QoS those.
But actual voice traffic travels over RTP protocol using port range specified per device, which would make it impossible to QoS that port range, as we don’t know which SIP port the RTP ports correspond to.
Basically with the way PAP2T handles VOIP, there is no way to direct one VOIP line’s traffic over one DSL connection, and other VOIP line’s to another.
Hoping I am making sense.
Sorry, my first answer went off on a little different topic (QoS on VoiP) while I think you were more worried about port mapping for RTP datagrams when you have multiple SIP devices behind a NATed interface.
In your “pre-boot” script is still your answer. This makes the Linux “net filter” (i.e. iptables) smart about SIP datagrams. It will look into the SIP exchange and determine that it needs to open and associate ports for RTP with the SIP session. It will also re-write the SIP datagrams to show the correct IP addresses and ports for the RTP ports that it opened, etc.
Basically it seems to make SIP through ZS “just work”. At least as far as port mapping and routing are concerned. You still should make sure your traffic classification and quality of service setup assures that your SIP/RTP streams are unaffected if/when other traffic is passing to your ISP.