From: Nick Matthews (matthn@gmail.com)
Date: Sun Jul 06 2008 - 21:09:55 ART
1) Dial peers, with the codec command. You'll want to set the incoming voip
dial peer in the case you described.
http://www.cisco.com/en/US/docs/ios/12_3/vvf_c/dial_peer/dp_confg.html#wp1067839
Something like
dial-peer voice 999 voip
incoming called-number .
codec g711ulaw
2) I don't think Cisco really relies on the RTCP, and frankly I wouldn't
trust it as much. I think the phones/gateways usually calculate all of this
themselves in another fashion, and is more reliable. You can check for lost
packets/jitter (and codec, duration, destination, etc) with 'show call
active voice brief'. Check out the xml description, and it should make
sense.
3) Between two direct endpoints the codec is almost always going to be the
same. There are ways to do one-way codecs, but it's very rarely done. What
that command is going to be telling you is what codec that DSP is using, and
I'd imagine for a call where they didn't match up (.000005% of calls) you
would use multiple DSPs and would see two different codecs.
This archive was generated by hypermail 2.1.4 : Mon Aug 04 2008 - 06:11:53 ART