I have a Cisco 871 router and a Cisco 7940G IP phone working directly with my ViaTalk line. Here's the VoIP LLQ config on my 871:
class-map match-any qos-call
match ip dscp ef
class-map match-any qos-call_setup
match ip dscp cs3
shape average 436000
service-policy output qos-out_fa4
What I'm not sure about is the priority/bandwidth values. Lets say 75 kbps is needed for the actual call (RTP media stream) and 16 kbps is needed for the SIP signalling for that call. If I'm going to use call waiting (i.e. I'm on a call and a call waiting call comes in), or if I'm 3-way calling, should both those values be doubled?
P.S. Does anyone by chance know how to change the default DSCP settings (for both RTP & SIP) on the 7940? Also, is there a way to set the RTP payload size?
I know how to set/remap DSCP settings on my 871. I figure why do it that way if there is a way to change the phone's default DSCP settings (24 for SIP; ef for RTP). What I really want to change is the DSCP for SIP.
Also, is there any way to change the phone's RTP packet size/duration? Default is 20ms/160 samples.
As for my 871 LLQ config, I'm still not sure what to base the priority and bandwidth values on for the two classes above. Again, my VSP is ViaTalk and my phone is registering directly with them. I've looked at Cisco's voice codec bw calc but none of the options seem to match my Internet connection type. I've got PPPoE ADSL with a DSL modem hanging off FA4 of my C871 router. PPPoE is configured on the C871.
These are the paths to get to each CCX logs through CLI. They may be helpful if you are having issues accessing RTMT or downloading logs through it.
If you want to download them you have to prefix "file get " and you can add one of the options (re...