ptime and voice/call quality problem
Hey Folks, Kind of strange issue here, tied to some other stuff I've posted about. Anyway, does anyone know how to set ptime or where that is configured? Please forgive if this is staring me in the face somewhere, I've not found anything satisfactory just looking around. But, I am slow sometimes. What's up? We have an FS that has two incoming numbers. One of them, just recently, started doing the following (from our VoIP ISP tech guy): All calls that work (to the second number) network originates the call with maxptime=20, we respond with ptime=20 and everything works fine. The calls that do not work (primary number) network originates the call with maxptime=30, we respond with ptime=30 then immediately re-invite with ptime=20. I suspect somewhere in this re-invite either your switch or our provider’s switch is getting confused and not sending or processing RTP based on the re-invite. Result? The caller can't be heard by the call taker at our FS. Plus, the sound quality, even of the ring, is bad. Sorry so long a post! Right now, we're forwarding the "bad" number to the "good". A band-aid. Any ptime experts here? We did not change this parameter, but did modify the call_timeout up to 19 from 15 trying to give 'em another ring. Then back down once this started to no avail. Thanks for reading this far if you did. Any advice appreciated.
Any chance you can get a SIP trace of working vs. not working calls? Put them on pastebin.freeswitch.org and the community will take a look. -MC On Fri, May 27, 2011 at 12:31 PM, Michael Gende <mgende@gendesign.com>wrote:
Hey Folks,
Kind of strange issue here, tied to some other stuff I've posted about. Anyway, does anyone know how to set ptime or where that is configured?
Please forgive if this is staring me in the face somewhere, I've not found anything satisfactory just looking around. But, I am slow sometimes.
What's up? We have an FS that has two incoming numbers. One of them, just recently, started doing the following (from our VoIP ISP tech guy):
All calls that work (to the second number) network originates the call with maxptime=20, we respond with ptime=20 and everything works fine. The calls that do not work (primary number) network originates the call with maxptime=30, we respond with ptime=30 then immediately re-invite with ptime=20. I suspect somewhere in this re-invite either your switch or our provider’s switch is getting confused and not sending or processing RTP based on the re-invite.
Result? The caller can't be heard by the call taker at our FS. Plus, the sound quality, even of the ring, is bad.
Sorry so long a post! Right now, we're forwarding the "bad" number to the "good". A band-aid. Any ptime experts here? We did not change this parameter, but did modify the call_timeout up to 19 from 15 trying to give 'em another ring. Then back down once this started to no avail.
Thanks for reading this far if you did. Any advice appreciated.
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Yes, I believe I can get some packet captures from our VoIP ISP. I'll post them to the group you suggest on Tuesday. Thanks for the suggestion! Mike G. On Fri, May 27, 2011 at 4:02 PM, Michael Collins <msc@freeswitch.org> wrote:
Any chance you can get a SIP trace of working vs. not working calls? Put them on pastebin.freeswitch.org and the community will take a look.
-MC
On Fri, May 27, 2011 at 12:31 PM, Michael Gende <mgende@gendesign.com>wrote:
Hey Folks,
Kind of strange issue here, tied to some other stuff I've posted about. Anyway, does anyone know how to set ptime or where that is configured?
Please forgive if this is staring me in the face somewhere, I've not found anything satisfactory just looking around. But, I am slow sometimes.
What's up? We have an FS that has two incoming numbers. One of them, just recently, started doing the following (from our VoIP ISP tech guy):
All calls that work (to the second number) network originates the call with maxptime=20, we respond with ptime=20 and everything works fine. The calls that do not work (primary number) network originates the call with maxptime=30, we respond with ptime=30 then immediately re-invite with ptime=20. I suspect somewhere in this re-invite either your switch or our provider’s switch is getting confused and not sending or processing RTP based on the re-invite.
Result? The caller can't be heard by the call taker at our FS. Plus, the sound quality, even of the ring, is bad.
Sorry so long a post! Right now, we're forwarding the "bad" number to the "good". A band-aid. Any ptime experts here? We did not change this parameter, but did modify the call_timeout up to 19 from 15 trying to give 'em another ring. Then back down once this started to no avail.
Thanks for reading this far if you did. Any advice appreciated.
_______________________________________________ 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
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Michael Collins
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Michael Gende