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

[Jamie Heckford]

We had very similar problems to this which drove us insane for ages.

Basically we use VoIP trunks (SIP) for all our inbound + outbound calls.
Call quality was good however we would get random problems where people
could not hear us or us hear them for about 5-10 seconds at a time.

After weeks of trying to get to the bottom of the problem it appeared
our VoIP trunk provider (sentiro/sip2go) had ropey connectivity at best.

We have since changed provider and now experience no call problems
whatsoever (after running extensive tests to the sip host such as mtr
etc.)

Jamie 

> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com 
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Matt
> Sent: 05 November 2006 13:30
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Audio goes one way during the 
> call for a fewseconds. Is it RTP, NAT, dyndns, or what it is?
> 
> 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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