asterisk users - Nov 2006 - Audio goes one way during the call for a few seconds. Is it RTP, NAT, dyndns, or what it is?

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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And to add to this, in scenarios 2 and 3, sometimes other party can listen
you but you can't listen them, i.e. opposite.
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Seems likes I am the only person in Asterisk world with this problem,
everybody else is fine with audio.
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Tried inspecting packet dumps with an analyser like Wireshark (ex
Ethereal)? They can prove very useful when troubleshooting issues like
these.

On 11/5/06, Zeeshan Zakaria <zishanov@gmail.com>
wrote:
> Seems likes I am the only person in Asterisk world with this problem,
> everybody else is fine with audio.
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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@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
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>