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Audio Delay
by stephen＠picardogroup.com 17 Aug '12

17 Aug '12

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Re: [Freeswitch-dev] Rare audio "clicks" inside mod_conference
by Peter Olsson 17 Aug '12

17 Aug '12

Thanks for the tip - it got me thinking in the right direction. I think I've found it now, and I'm running my changed code in production for some testing. If everything looks good I'll post a patch on Jira after the weekend. /Peter -----Ursprungligt meddelande----- Från: freeswitch-dev-bounces(a)lists.freeswitch.org [mailto:freeswitch-dev-bounces@lists.freeswitch.org] För Anthony Minessale Skickat: den 15 augusti 2012 18:45 Till: freeswitch-dev(a)lists.freeswitch.org Ämne: Re: [Freeswitch-dev] Rare audio "clicks" inside mod_conference Add some debugging to the places in switch_core_io.c that do memset, maybe its from that. On Wed, Aug 15, 2012 at 7:57 AM, Peter Olsson <peter.olsson(a)visionutveckling.se> wrote: > Before anyone mentions it (since I'm usually one of those who does), I > will submit this to Jira, but I would appreciate some more input > before doing so > :) > > > > /Peter > > > > Från: freeswitch-dev-bounces(a)lists.freeswitch.org > [mailto:freeswitch-dev-bounces@lists.freeswitch.org] För Peter Olsson > Skickat: den 15 augusti 2012 14:50 > Till: freeswitch-dev(a)lists.freeswitch.org > Ämne: [Freeswitch-dev] Rare audio "clicks" inside mod_conference > > > > I'm not really sure where to ask about this, but I'll give it a try on > the dev-list for starters.. > > > > I sometimes (very rarely, but often enough to want to find the cause) > get audio "clicks" in the audio stream on devices connected to a FS conference. > I have samples from wireshark showing the problem. In L16 format it > seems there are a few bytes set to 32767 (max int16 value), that is > causing the actual audio. > > > > The strange thing is that when this occurs, there is actually no one > that sends this audio (but all members except one will get this audio, > so it has been put into the buffer by the member without the click), > so I'm starting to think if this is a generated packet inside FS, that > is not initialized correctly, or if the decoding from PCMA/PCMU to L16 fails for some reason. > > > > I've traced it down so far that I know how to handle it (at least work > around it), but I don't know how to reproduce it, and I don't know the > real cause. With the help of lots of debugging I've seen that this > occurs when a partial frame is read from a member (however, according > to wireshark, this partial frame was never sent over the network). We > have 8khz conferences, and the connected members use PCMA and/or PCMU, > so normally when audio from a member is appended to the audio buffer, > the length of the data is 320 bytes (I guess this is L16). When the > click occurs, I've seen that the datasize is only 160 bytes instead, > which would more indicate a undecoded PCMA/PCMU frame. And as I said, > there is no packet with half the payload size anywhere in wireshark, > so I'm not really sure where this is coming from. I've looked through > most of the code in switch_rtp.c and mod_sofia.c (that's related to RTP), but I can't really find any good reason. > > > > Right now I've added a check (if read_frame->datalen != > member->read_impl.encoded_bytes_per_packet), and when using this it > member->works as > expected (at least so far - as I said, it happens rarely), but I guess > I might miss a frame of audio. I will try to look into this further, > but if anyone have any suggestions I'm open for ideas :) > > > > /Peter > > > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-dev mailing list FreeSWITCH-dev(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org !DSPAM:502bcf8532761778320602!

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Re: [Freeswitch-dev] Audio Delay
by Peter Olsson 17 Aug '12

17 Aug '12

Nope... :) Are you on real hardware, or a virtual environment? Try issuing “timer_test” in the console or the fs_cli, check if the results look ok – usually this is a timing problem, but there are lots of other possibilities – network, hardware problems etc. /Peter Från: freeswitch-dev-bounces(a)lists.freeswitch.org [mailto:freeswitch-dev-bounces@lists.freeswitch.org] För stephen(a)picardogroup.com Skickat: den 17 augusti 2012 05:11 Till: freeswitch-dev(a)lists.freeswitch.org Ämne: [Freeswitch-dev] Audio Delay Cent OS 5.4, Intel I7, FS 1.2 Latest release , sangoma 101d card with latest driver (wanpipe) PRI circuit. When two people in conference, we are experiencing increased audio delay as call goes longer. Example in a 30 min time period after just 4-5 mins, the delay gets longer and longer reaching almost 1 minute after 10 mins or so. Anyone else experienced this problem? !DSPAM:502ddf1b32761645758032!

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Audio Delay
by stephen＠picardogroup.com 17 Aug '12

17 Aug '12

1 0

Mod_Callcenter queue and hashtable
by piyush singhai 16 Aug '12

16 Aug '12

Hello, I am trying to change the code in mod_callcenter. I want to implement queue add functionality in the code currently it is not available. i followed this link http://jira.freeswitch.org/browse/FS-3338 to add a patch in the mod_callcenter.c file and add the queue also in the database. I have some questions while following the code. Please correct me if my understanding is wrong. 1. There is no database table for queue that is created in the xml and after applying the patch also. after following the code i got that it is stored in the hashtable. So why i can not see in the database and why it is storing in the hashtable.? How mod_Callcenter storing queue information in the freeswitch. 2. Is there any other way to add queue at runtime.?? 3. what is stored in the members table? Please help me out.

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Re: [Freeswitch-dev] Rare audio "clicks" inside mod_conference
by Peter Olsson 15 Aug '12

15 Aug '12

Before anyone mentions it (since I'm usually one of those who does), I will submit this to Jira, but I would appreciate some more input before doing so :) /Peter Från: freeswitch-dev-bounces(a)lists.freeswitch.org [mailto:freeswitch-dev-bounces@lists.freeswitch.org] För Peter Olsson Skickat: den 15 augusti 2012 14:50 Till: freeswitch-dev(a)lists.freeswitch.org Ämne: [Freeswitch-dev] Rare audio "clicks" inside mod_conference I'm not really sure where to ask about this, but I'll give it a try on the dev-list for starters.. I sometimes (very rarely, but often enough to want to find the cause) get audio "clicks" in the audio stream on devices connected to a FS conference. I have samples from wireshark showing the problem. In L16 format it seems there are a few bytes set to 32767 (max int16 value), that is causing the actual audio. The strange thing is that when this occurs, there is actually no one that sends this audio (but all members except one will get this audio, so it has been put into the buffer by the member without the click), so I'm starting to think if this is a generated packet inside FS, that is not initialized correctly, or if the decoding from PCMA/PCMU to L16 fails for some reason. I've traced it down so far that I know how to handle it (at least work around it), but I don't know how to reproduce it, and I don't know the real cause. With the help of lots of debugging I've seen that this occurs when a partial frame is read from a member (however, according to wireshark, this partial frame was never sent over the network). We have 8khz conferences, and the connected members use PCMA and/or PCMU, so normally when audio from a member is appended to the audio buffer, the length of the data is 320 bytes (I guess this is L16). When the click occurs, I've seen that the datasize is only 160 bytes instead, which would more indicate a undecoded PCMA/PCMU frame. And as I said, there is no packet with half the payload size anywhere in wireshark, so I'm not really sure where this is coming from. I've looked through most of the code in switch_rtp.c and mod_sofia.c (that's related to RTP), but I can't really find any good reason. Right now I've added a check (if read_frame->datalen != member->read_impl.encoded_bytes_per_packet), and when using this it works as expected (at least so far - as I said, it happens rarely), but I guess I might miss a frame of audio. I will try to look into this further, but if anyone have any suggestions I'm open for ideas :) /Peter !DSPAM:502b98ee32764854613503!

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Weekly FreeSWITCH Conference Call
by Ken Rice 15 Aug '12

15 Aug '12

Hello Gang! This weeks conf call will be starting soon... The Agenda Page for today: http://wiki.freeswitch.org/wiki/FS_weekly_2012_08_15 I will be talking about FreeSWITCH 1.2, how to get it and whats news. We¹ll follow this up with community discussion! Join us! Ken

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Rare audio "clicks" inside mod_conference
by Peter Olsson 15 Aug '12

15 Aug '12

I'm not really sure where to ask about this, but I'll give it a try on the dev-list for starters.. I sometimes (very rarely, but often enough to want to find the cause) get audio "clicks" in the audio stream on devices connected to a FS conference. I have samples from wireshark showing the problem. In L16 format it seems there are a few bytes set to 32767 (max int16 value), that is causing the actual audio. The strange thing is that when this occurs, there is actually no one that sends this audio (but all members except one will get this audio, so it has been put into the buffer by the member without the click), so I'm starting to think if this is a generated packet inside FS, that is not initialized correctly, or if the decoding from PCMA/PCMU to L16 fails for some reason. I've traced it down so far that I know how to handle it (at least work around it), but I don't know how to reproduce it, and I don't know the real cause. With the help of lots of debugging I've seen that this occurs when a partial frame is read from a member (however, according to wireshark, this partial frame was never sent over the network). We have 8khz conferences, and the connected members use PCMA and/or PCMU, so normally when audio from a member is appended to the audio buffer, the length of the data is 320 bytes (I guess this is L16). When the click occurs, I've seen that the datasize is only 160 bytes instead, which would more indicate a undecoded PCMA/PCMU frame. And as I said, there is no packet with half the payload size anywhere in wireshark, so I'm not really sure where this is coming from. I've looked through most of the code in switch_rtp.c and mod_sofia.c (that's related to RTP), but I can't really find any good reason. Right now I've added a check (if read_frame->datalen != member->read_impl.encoded_bytes_per_packet), and when using this it works as expected (at least so far - as I said, it happens rarely), but I guess I might miss a frame of audio. I will try to look into this further, but if anyone have any suggestions I'm open for ideas :) /Peter

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FreeSWITCH 1.2.1 has been rolled...
by Ken Rice 14 Aug '12

14 Aug '12

FreeSWITCH 1.2.1 is Here!!! Tarball is on files.freeswitch.org as usual. Also Debian users, check out the debian repo on http://files.freeswitch.org/repo More info on this coming soon, currently this continues nightly builds from the dev branch but will soon also contain a stable repo... As usual, any issues please report them via jira.freeswitch.org and please tag them with the correct version! K

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Level3 and 25s failover
by Tom Parrott 11 Aug '12

11 Aug '12

Hi, The company I work for get toll-free numbers from Level3 via Bandwidth.com which are terminated on several Freeswitch servers using DNS SRV records. Annoyingly Level3 have a 'feature' where they will failover a call to a different server if the call is not answered in 25 seconds. In the UK, the provider we use has a similar failover setting, but as soon as early media is sent the fail-over mechanism is stopped and you can ring for as long as you need to. I have been told by Level3 that it is a known issue, but not something they can/will change, and that calls should be answered within 25 seconds. I tend to agree, however I forward calls to our customer's PSTN numbers, and they do not always answer within that time. This causes duplicate call records when the call fails over. For a call analytics company, this causes serious problems in statistics. Does anyone have any experience with Level3 and this strange behaviour? Thanks Tom

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