asterisk users - Apr 2005 - About Audio Latency from PSTN to SIP

Hi all,

I built a VoIP PBX box with asterisk and one x100p card. Every thing is ok
except there is a short audio latency from PSTN to SIP and no delay in the
reverse direction.

At the beginning of a call, the latency is not very obvious, and it becomes more
and more obvious along with time. If the call keep 10 minutes, the delay will be
about half or one second.

Any suggestion?

--- 
Best regards,
 Qiao Yuansong
 mailto: qys@iscas.ac.cn
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HI,
 
Anybody here used SRV feature in Asterisk to route sip calls? If anybody
were able to get this working, I like to know how the sip configuration was
set. I am testing this in a lab with two different domain. I was hoping
Asterisk will be able to route the call by doing DNS look up on SRV. My lab
test if failing. Let me know if anybody here did it.
 
Thanks
 
&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*&*
C.M. Rahman Jr.
CCNP, MCSE Security    "Secure your self by securing your System"
CompTI Security Plus Certified
CCS Internet
 <http://www.ccsi.com/> http://www.ccsi.com
13706 Research Blvd. Suite 100
Austin, TX 78750
Tel: 512-257-2274 Ex: 115
  _____  

 
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Hello list,

Does anyone know about a web/php interface to deal with users in Realtime's 
Mysql database (sipusers and sippeers tables) ?

Thanks in advance

Laurent


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Hi all,

I built a VoIP PBX box with asterisk and one x100p card. Every thing is ok
except there is a short audio latency from PSTN to SIP and no delay in the
reverse direction.

At the beginning of a call, the latency is not very long, but it becomes more
and more obvious along with time. If the call keep 10 minutes, the delay will be
about half or one second.

Anyone knows the reason, and any suggestion?

Thanks a lot.

--- 
Best regards,
 Qiao Yuansong
 mailto: qys@iscas.ac.cn
-------------- next part --------------
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Hi all,

I built a VoIP PBX box with asterisk and one x100p card. Every thing is ok
except there is a short audio latency from PSTN to SIP and no delay in the
reverse direction.

At the beginning of a call, the latency is not very long, but it becomes more
and more obvious along with time. If the call keep 10 minutes, the delay will be
about half or one second.

Anyone knows the reason, and any suggestion?

Thanks a lot.

--- 
Best regards,
 Qiao Yuansong
 mailto: qys@iscas.ac.cn
-------------- next part --------------
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Hi all,

I built a VoIP PBX box with asterisk and one x100p card. Every thing is ok
except there is a short audio latency from PSTN to SIP and no delay in the
reverse direction.

At the beginning of a call, the latency is not very long, but it becomes more
and more obvious along with time. If the call keep 10 minutes, the delay will be
about half or one second.

Anyone knows the reason, and any suggestion?

Thanks a lot.

--- 
Best regards,
 Qiao Yuansong
 mailto: qys@iscas.ac.cn
-------------- next part --------------
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--- Qiao Yuansong <qys@iscas.ac.cn> wrote:
> At the beginning of a call, the latency is not very
> long, but it becomes more and more obvious along
> with time. If the call keep 10 minutes, the delay
> will be about half or one second.
> 
> Anyone knows the reason, and any suggestion?
Are you running your asterisk from behind a nat?  My
asterisk is behind a nat and I have the same problem
with iax. Two other guys includong Mr. Andrew Kohsmith
has the same issue and he is working on this problem.
Today, he sent me an ip address to dial  where he had
echo test. The RTT (ping round trip time) from iax was
low and the same almost all the time and the jitter
was down to zero to this his server.

the high jitter (variation in packet delay) causing a
compounding problem that eventually cause the
communication break down.The question becomes what is
the source of the jitter: network layer, low internet
bandwidth or some iax.conf or sip.conf configuration




		
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