[sip-comm] Re: long delay when initiating RTP session with SIP and asterisk


#1

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 as 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

···

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#2

Hi,
things you can try is get the jmf version from sip-communicator svn and try running with it.
I was also surprised after setting everything to ipaddresses that there was an issue with dns lookups in jmf.
The one slow down you can reproduce it yourself try constructing InetAddress from "0.0.0.0" which is the bind address
an try - inetaddress.getHostName(). Is it slow?
There was also one more similar problem in the fmj RTPSessionMgr.

damencho

···

FRANKOFF.S@forces.gc.ca wrote:

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 as 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,
  
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#3

Hi!
If Asterisk is configured for the routing of the RTP flows, it always tries to update the new sessions according to its codecs using some RE-INVITE mechanism. If your Asterisk is so configured, could be there some problem with JAIN/JMF interwork ? ( because of the slowness of JAIN/JMF )
Is Asterisk configured for the routing of RTP? Maybe, changing the Asterisk setting you could solve your problem.

/BR Paolo

···

----- Original Message -----
  From: Damian Minkov
  To: users@sip-communicator.dev.java.net
  Sent: Wednesday, September 19, 2007 4:14 PM
  Subject: Re: [sip-comm] Re: long delay when initiating RTP session with SIP and asterisk

  Hi,
  things you can try is get the jmf version from sip-communicator svn and
  try running with it.
  I was also surprised after setting everything to ipaddresses that there
  was an issue with dns lookups in jmf.
  The one slow down you can reproduce it yourself try constructing
  InetAddress from "0.0.0.0" which is the bind address
  an try - inetaddress.getHostName(). Is it slow?
  There was also one more similar problem in the fmj RTPSessionMgr.

  damencho

  FRANKOFF.S@forces.gc.ca wrote:
  > 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 as 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,
  >

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