[asterisk-users] Softphone RTP Session Start-up Delay

[Kutman.DK at forces.gc.ca]

Hi,

Thanks very much for your reply.  I would like to add some information which may provide a little more clarification on this matter.  The LAN network that we presently have consists of the Asterisk PC and two User PC's (This network is not connected to the internet).  To confirm that Asterisk/Trixbox operated correctly we installed an X-Lite phone on each user pc.  We specified the IP Address of the Asterisk machine for the domain in the properties of the X-Lite.  These X-Lites worked well, having no delay at any point in the process from when the call is made, up to the audio conversation.  Unfortunately, the X-Lite phone is not open-source, so we do not have the code available to us.  We then obtained the Jain-SIP phone, which is an open-source SIP softphone.  As done in the X-Lite, the Asterisk IP Address is specified for the "outbound proxy" or the domain.  We are now able to establish an audio conversation except for the fact that the RTP session takes about 20 seconds to setup, as mentioned before.  I am not sure if the DNS issue comes into play here because we are actually specifying the IP Address of the Asterisk server, but I am willing to try anything to fix this problem.  The two user pc's are setup on workgroups, so I do not believe that there is a domain available that can be entered in the hosts file.  Could the DNS still be the issue?  If not, would anyone be able to suggest any other possible problems that may be causing this delay.

Thanks in advance for the help,

Denis


-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com]On Behalf Of Anselm
Martin Hoffmeister
Sent: Monday, September 17, 2007 4:36 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Softphone RTP Session Start-up Delay


Am Montag, den 17.09.2007, 15:50 -0400 schrieb Kutman.DK at forces.gc.ca:
> Hello,
> 
> I have a small LAN network where I am running a Jain-Sip softphone on two user pc's.
> These softphones are connected through Asterisk(Trixbox).  Although the phones do
> work in providing an audio conversation, there is a long delay(about 20 seconds)
> in the initial RTP session setup.  I have tried a few values for the buffer length
> including setting it to zero.  I assumed this would drastically reduce the delay
> but there was no change.  I also tried a number of values for the minimum threshold
> and this did not change the amount of delay either.  Would anyone have an idea of
> why this delay is occurring and possibly how to reduce it?  

Hello Denis,

delays in that magnitude (20 seconds or about) may be related to DNS
issues - like trying to resolve a hostname, or trying to find a hostname
for an IP address. You could try to add all relevant IPs to
the /etc/hosts file (or C:\windows\system32\drivers\etc\hosts), like

192.168.0.2 host2

and see wether that helps.

Regards,
Anselm


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