Hey guys,
I need some assistance in tracking down the cause of audio problems that
are occurring at two of my sites:
Both sites run Asterisk 1.2.10 and use Sangoma A101u PRI cards. Both
sites are reporting that audio in calls is "dropping out" during
words,
so that the other caller (i.e. the remote user) can only hear bits of
the words.
This used to only happen on Asterisk-to-Asterisk calls via IAX2 (using
g729) so I assumed it was latency or bandwidth problems on the
inter-office network. However, the network is hardly used and my
round-trip times are sub 100ms according to iax2 show peers (with
qualify=yes).
Then, thinking it might be g729 issues, I changed the entire system to
only use alaw and the problem persists.
Does anyone have any suggestions on where to look next? My users are
getting increasingly annoyed and I'm quickly running out of ideas.
Thanks,
Avi
--
National Manager - Special Projects
< Sydney / Melbourne / Canberra / Hobart / London />
2/340 Gore Street T: 1 3000 SQUIZ (77849)
Fitzroy, VIC T: +61 (0) 3 9235 5400
3065 F: +61 (0) 3 9235 5444
W: http://www.squiz.net/
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Avi Miller wrote:> Does anyone have any suggestions on where to look next? My users are > getting increasingly annoyed and I'm quickly running out of ideas.Replying to myself to note that this is now happening on outbound calls via ISDN, i.e. calls that don't use IAX2 or the inter-office network. It also happens on inbound calls. Ta, Avi -- National Manager - Special Projects < Sydney / Melbourne / Canberra / Hobart / London /> 2/340 Gore Street T: 1 3000 SQUIZ (77849) Fitzroy, VIC T: +61 (0) 3 9235 5400 3065 F: +61 (0) 3 9235 5444 W: http://www.squiz.net/ .....>> Open Source - Own it - Squiz.net ...../>
Avi, We need more info, Through what means are both sides connected, 1:1 xDSL? What bandwidth, are you using tunnels (pptp/gre/ipsec), how many concurrent calls etc. You could try analysing network delay/jitter/packetloss using Smokeping. Note that on DSL 1 g729 calls uses about 45 kbit/s, alaw uses about 108 kbit on DSL Erik Avi Miller wrote:> Hey guys, > > I need some assistance in tracking down the cause of audio problems that > are occurring at two of my sites: > > Both sites run Asterisk 1.2.10 and use Sangoma A101u PRI cards. Both > sites are reporting that audio in calls is "dropping out" during words, > so that the other caller (i.e. the remote user) can only hear bits of > the words. > > This used to only happen on Asterisk-to-Asterisk calls via IAX2 (using > g729) so I assumed it was latency or bandwidth problems on the > inter-office network. However, the network is hardly used and my > round-trip times are sub 100ms according to iax2 show peers (with > qualify=yes). > > Then, thinking it might be g729 issues, I changed the entire system to > only use alaw and the problem persists. > > Does anyone have any suggestions on where to look next? My users are > getting increasingly annoyed and I'm quickly running out of ideas. > > Thanks, > Avi >