FS "Holds On" to ITSP's 183 for 2 Sec.Before Relaying to Phone?
Hi, Just tinkering around with FreeSWITCH pretty much using the default configuration, with some customization to connect my Polycom 430 and my ITSP. The problem I am having is that the first few seconds of calls (or early media if available) have no audio. When I NGREP'd the SIP dialog between my Phone <> FS and FS <> ITSP with timestamps, I can see that FreeSwitch is holding onto the 183 and subsequent 200 OK for about 2 seconds before relaying it onto the Polycom. FWIW, server is an Intel Atom 330. Any idea why this is... I am sure it must be a configuration issue somewhere? I did not have this issue with Asterisk, so I am hoping to troubleshoot it and move on with FreeSWITCH Thanks. AF.
Alan, Welcome to FreeSWITCH, Lets start out with some questions... Is FreeSWITCH behind nat? Can you type "sofia status profile internal" and reply with that? /b On Jun 4, 2010, at 11:12 AM, Alan Frisch wrote:
Hi,
Just tinkering around with FreeSWITCH pretty much using the default configuration, with some customization to connect my Polycom 430 and my ITSP. The problem I am having is that the first few seconds of calls (or early media if available) have no audio.
When I NGREP'd the SIP dialog between my Phone <> FS and FS <> ITSP with timestamps, I can see that FreeSwitch is holding onto the 183 and subsequent 200 OK for about 2 seconds before relaying it onto the Polycom. FWIW, server is an Intel Atom 330.
Any idea why this is... I am sure it must be a configuration issue somewhere? I did not have this issue with Asterisk, so I am hoping to troubleshoot it and move on with FreeSWITCH
Thanks.
AF.
type the following at the cli console loglevel debug sofia profile internal siptrace on repeat and post the trace to http://pastebin.freeswitch.org (pass is in the dialog box) On Fri, Jun 4, 2010 at 11:12 AM, Alan Frisch <frisch.alan@gmail.com> wrote:
Hi,
Just tinkering around with FreeSWITCH pretty much using the default configuration, with some customization to connect my Polycom 430 and my ITSP. The problem I am having is that the first few seconds of calls (or early media if available) have no audio.
When I NGREP'd the SIP dialog between my Phone <> FS and FS <> ITSP with timestamps, I can see that FreeSwitch is holding onto the 183 and subsequent 200 OK for about 2 seconds before relaying it onto the Polycom. FWIW, server is an Intel Atom 330.
Any idea why this is... I am sure it must be a configuration issue somewhere? I did not have this issue with Asterisk, so I am hoping to troubleshoot it and move on with FreeSWITCH
Thanks.
AF.
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
Brian/Anthony, Looks like I forgot to remove the <-- and --> from around the custom RTP settings. This may have been causing the hang and calls are now getting full early media, without cut-offs. Thanks and have a great weekend! AF. On Fri, Jun 4, 2010 at 12:56 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
type the following at the cli console loglevel debug sofia profile internal siptrace on repeat and post the trace to http://pastebin.freeswitch.org (pass is in the dialog box)
On Fri, Jun 4, 2010 at 11:12 AM, Alan Frisch <frisch.alan@gmail.com> wrote:
Hi,
Just tinkering around with FreeSWITCH pretty much using the default configuration, with some customization to connect my Polycom 430 and my ITSP. The problem I am having is that the first few seconds of calls (or early media if available) have no audio.
When I NGREP'd the SIP dialog between my Phone <> FS and FS <> ITSP with timestamps, I can see that FreeSwitch is holding onto the 183 and subsequent 200 OK for about 2 seconds before relaying it onto the Polycom. FWIW, server is an Intel Atom 330.
Any idea why this is... I am sure it must be a configuration issue somewhere? I did not have this issue with Asterisk, so I am hoping to troubleshoot it and move on with FreeSWITCH
Thanks.
AF.
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
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On Fri, Jun 4, 2010 at 10:28 AM, Alan Frisch <frisch.alan@gmail.com> wrote:
Brian/Anthony,
Looks like I forgot to remove the <-- and --> from around the custom RTP settings. This may have been causing the hang and calls are now getting full early media, without cut-offs.
Thanks and have a great weekend!
Hehe, welcome to XML. :) You'll be used to it in no time. Just be sure to use a text editor with syntax highlighting. -MC
participants (4)
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Alan Frisch
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Anthony Minessale
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Brian West
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Michael Collins