[Asterisk-Users] rtp sound quality?
pstn -> sip gw -> * -> C7960 When I dial into * via the pstn, I hear the ivr menu just fine (good quality). I press 3000 (valid extn), and I begin to hear ringing however the ring back is very very choppy. I answer the C7960, and speech is clear in both directions. Place the C7960 extn on hold, and the MOH is very choppy. Checking 'sip show channels' indicates both the sip gw and C7960 are using ulaw; Both are 100 meg on local LAN. Top suggests all processes running less then 1 or 2 percent. The choppy sound only happens on the sip gw (Mediatrix 1204). MOH between two C7960's works fine. MOH via x100p works fine. Tried canreinvite=no and yes; no difference. Using a packet sniffer, I see a ~200 millisecond delay about every 1/2 second or so (varys), but nothing within the trace to hint at a layer-2 problem. Anyone have any thoughts as to why ringback and MOH are choppy but conversations are fine? Anything else I can look at to isolate the issue? Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] rtp sound quality?
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson >pstn -> sip gw -> * -> C7960 > >When I dial into * via the pstn, I hear the ivr menu just fine (good >quality). I press 3000 (valid extn), and I begin to hear ringing however >the ring back is very very choppy. Where are you getting timing from? Zaptel device? Ztdummy? -Josh ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] rtp sound quality?
> >pstn -> sip gw -> * -> C7960 > > > >When I dial into * via the pstn, I hear the ivr menu just fine (good > >quality). I press 3000 (valid extn), and I begin to hear ringing > however >the ring back is very very choppy. > > Where are you getting timing from? Zaptel device? Ztdummy? The * system has a pair of x100p's installed (and working), so I'm "assuming" zaptel (unless timing requires the call to come through the x100p first). Using the MOH (as an example), music is very choppy; however, I've noticed that if I try to talk over the top of MOH, then MOH sounds fine. Is this really the old timing thingie here too? Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] rtp sound quality?
Rich Adamson wrote: pstn -> sip gw -> * -> C7960 When I dial into * via the pstn, I hear the ivr menu just fine (good quality). I press 3000 (valid extn), and I begin to hear ringing however the ring back is very very choppy. I answer the C7960, and speech is clear in both directions. Place the C7960 extn on hold, and the MOH is very choppy. Checking 'sip show channels' indicates both the sip gw and C7960 are using ulaw; Both are 100 meg on local LAN. Top suggests all processes running less then 1 or 2 percent. The choppy sound only happens on the sip gw (Mediatrix 1204). MOH between two C7960's works fine. MOH via x100p works fine. Tried canreinvite=no and yes; no difference. Using a packet sniffer, I see a ~200 millisecond delay about every 1/2 second or so (varys), but nothing within the trace to hint at a layer-2 problem. Anyone have any thoughts as to why ringback and MOH are choppy but conversations are fine? Anything else I can look at to isolate the issue? You need to disable VAD on the 1204. The 1204 stops xmiting RTP to * if it does not detect any acoustic energy. * can not clock itself sending RTP packets. It relyes on receiving RTP packets for it's timing. Try singing along with your MOH and the choppiness should go away, or disable VAD, or fix * RTP driver. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] rtp sound quality?
> >pstn -> sip gw -> * -> C7960 > > > >When I dial into * via the pstn, I hear the ivr menu just fine (good > >quality). I press 3000 (valid extn), and I begin to hear ringing however the > >ring back is very very choppy. > > > >I answer the C7960, and speech is clear in both directions. Place the C7960 > >extn on hold, and the MOH is very choppy. Checking 'sip show channels' indicates > >both the sip gw and C7960 are using ulaw; Both are 100 meg on local LAN. Top > >suggests all processes running less then 1 or 2 percent. > > > >The choppy sound only happens on the sip gw (Mediatrix 1204). MOH between two > >C7960's works fine. MOH via x100p works fine. > > > >Tried canreinvite=no and yes; no difference. Using a packet sniffer, I see a > >~200 millisecond delay about every 1/2 second or so (varys), but nothing within > >the trace to hint at a layer-2 problem. > > > >Anyone have any thoughts as to why ringback and MOH are choppy but conversations > >are fine? Anything else I can look at to isolate the issue? > > > You need to disable VAD on the 1204. > The 1204 stops xmiting RTP to * if it does not detect any acoustic energy. > > * can not clock itself sending RTP packets. > It relyes on receiving RTP packets for it's timing. > Try singing along with your MOH and the choppiness should go away, or > disable VAD, or fix * RTP driver. Thanks Bob, that fixed it. Any other hints/issues/default values that I should muck with, or is that about it? Seems like it works pretty good; excellent echo cancellation, etc. I haven't done anything with the box as yet for dialing outbound. Anything to be concerned with, special parameters, etc? Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] rtp sound quality?
Hi Rich! > Anyone have any thoughts as to why ringback and MOH are choppy but > conversations are fine? Anything else I can look at to isolate the > issue? First guess (rather likely): Silence supression Second guess (unlikely): Non optimal "Voice frames per TX" as it is called in Grandstream setup; don't have a Mediatrix, so I can only guess. Should be 2 for the GS. Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] rtp sound quality?
> > Anyone have any thoughts as to why ringback and MOH are choppy but > > conversations are fine? Anything else I can look at to isolate the > > issue? > > First guess (rather likely): Silence supression > > Second guess (unlikely): Non optimal "Voice frames per TX" as it is > called in Grandstream setup; don't have a Mediatrix, so I can only guess. > Should be 2 for the GS. Philipp, First guess was right on! Sounds great now. Working on dialout now. Can't seem to get the syntax right so far. But, will get there. :) Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] rtp sound quality?
Rich Adamson wrote: Thanks Bob, that fixed it. Any other hints/issues/default values that I should muck with, or is that about it? Seems like it works pretty good; excellent echo cancellation, etc. I haven't done anything with the box as yet for dialing outbound. Anything to be concerned with, special parameters, etc? I can't think of anything off the top of my head. It has been a while since I set mine up. My one and only complaint so far with this box is the snmp config stuff. They only give you a windows version. I have no windows boxes in my office. I just thought some day I would have to slam together a few little snmp scripts or gui code that drives off their MIB files. But I never had to go back into the box to do anything, so this has been a low priority. I am just a low level c hack. Before I go out and write any thing to do this snmp admin stuff, are there any linux tools I could use to do this? -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
