[asterisk-users] Audio goes one way during the call for a few seconds. Is it RTP, NAT, dyndns, or what it is?

[Matt]

Sounds like a bad Internet connection messing with the IAX
jitterbuffer.  Try running ping plotter from your location to your
host, and see if it goes 'red'/down.

On 11/3/06, Zeeshan Zakaria <zishanov at gmail.com> wrote:
> Hi everybody,
>
> I finally want to get rid of 1-way audio problem. Please help me here.
>
> I have 3 scenarios.
>
> 1. Audio is always one way. Caller who dialed can't listen the called party
> but called party can listen him. In this scenatio Asterisk is on dynamic IP
> with dyndns FQDN. sip.conf has externip = abc.dyndns.org and localnet =
> xxx.xxx.xxx.xxx entry. Trunk and extensions are SIP. Where is the voice
> getting lost from the called party? NAT is there but Asterisk is in DMZ.
>
> 2. Conversation is going fine when all of a sudden you realize that other
> parth has started saying 'hello, hello' because they can't hear you. But you
> are hearing them loud and clear. Now you are on static IP with dyndns FQDN.
> externip and localnet settings in sip.conf (do we need them for static IP?).
> After about 15-20 seconds, again 2-way converstaion is established again.
> IAX trunk, SIP extension, no NAT.
>
> 3. Conversation goes one way for 15-20 sec during the most important part of
> the conversation (Murphy's Law). You are on a static IP with no dyndns
> enrty. Trunk is ZAP on PRI, extensions SIP. NAT present but router properly
> configures for port forwarding. externip and localnet settings present in
> sip.conf
>
> Is think may be due to some reason RTP stream gets lost, routed to wrong IP.
> But why would this happen during a call and how to stop it from happening.
> Or is there some other reason behind this? Does dyndns setting have to do
> anything with this problem? How can I overcome this problem once and
> forever.
>
> --
> Zeeshan A Zakaria
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