Posted: Mon Jun 01, 2009 5:20 am Post subject: [asterisk-users] Suddenly the voice became garbage (like rob
Michelle Dupuis escribió:
You're not alone...we never found the cause of this (rare) occurance...
[mailto:firstname.lastname@example.org] On Behalf Of bilal ghayyad
Sent: Sunday, May 31, 2009 8:58 PM
To: Asterisk Users List
Subject: [asterisk-users] Suddenly the voice became garbage (like
robot)using Asterisk 188.8.131.52
I was using since one year Asterisk 184.108.40.206 and zaptel 220.127.116.11 and they
were working fine via SIP, IAX and Digium fxo and fxs ports.
Suddenly just before 2 or 3 days, the voice become garbage like robot when I
place a call from the SIP Phone (which is in a country and the Asterisk box
in another country). I am surprise what is the reason that let rtp become
like this ! The sound now like robot (human machine:)--
What could be? Actually before about 10 days, we added one Polycom SIP Phone
and added qualify=yes for the SIP entities, also I fixed the externip to be
the public IP address of the machine, and I fixed the localnet (I was not
using externip and localnet, because I was not need them for NAT issues, but
I used it when I start have NATed Polycom devices). But even, the voice was
fine after these changes, so I do not think it has any relation.
Could it be related the Asterisk 18.104.22.168 and how it manipulated the rtp,
packets, so maybe a small changes in the Internet provider effected and let
things becoming bad??!!
It sounds like an ISP problem. Is your bandwidth ok? Do you share the
connection with another services? In my case, usually "robot" calls are
related to upload congestion, and implementing egress QoS policies on
the router helped a lot to maintain the good quality of the calls. This
may be your case.
If your asterisk worked fine for one entire year flawlessly, you surely
can't blame it for bad call quality.
I used gsm and g729 codecs, and both are bad. g729 give little bit better
quality than gsm (and both are bad now .. garbage). The voice becoming bad
even if we do a call using IAX Trunk or from SIP Phone (Polycom).
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