From: Bit Gossip (bit.gossip@chello.nl)
Date: Sun Jul 06 2008 - 15:21:28 ART
Experts,
I am not sure if this is the right list for voip as well, if not can you
please advise a nice list/group where I can post it?
Anyway I have setup a very simple voip gateway on my cisco 2621 running
c2600-is-mz.123-26.bin with c2600-is-mz.123-26.bin and attached to the
internet via the FE interfaces.
People can call me on the public interface and all works fine: my pots rings
and we can talk....
1) how can I force a certain type of codec?
2) when i do: 'debug voice rtcp' what I see in the lost packet is: how many
packets that my interlocutor has missed out of my RTP stream?
3) if there is a working call setup between A and B and on A I type:
A#show voice dsp
DSP DSP DSPWARE CURR BOOT PAK TX/RX
TYPE NUM CH CODEC VERSION STATE STATE RST AI VOICEPORT TS ABORT PACK
COUNT
==== === == ======== ======= ===== ======= === == ========= == =====
============
C542 001 01 gsmfr 4.1.46 busy idle 0 0 1/0/0 NA 0
22844/80407
the codec gsmfr is from A to B or from B to A? Or the codec is always the same
for the two directions of the calls?
Thanks,
Bit.
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
voice service pots
!
voice service voip
sip
!
voice class codec 2
codec preference 1 gsmfr
!
voice-port 1/0/0
!
voice-port 1/0/1
!
!
!
!
dial-peer cor custom
!
!
!
dial-peer voice 225 pots
destination-pattern 225
port 1/0/0
This archive was generated by hypermail 2.1.4 : Mon Aug 04 2008 - 06:11:53 ART