autofix incorrect packetization
Hi, I get choppy audio in conferences from endpoints that send G722@30mspackets. The endpoint in question is a FRITZ!Box CPE and this seems to be a known problem with these type of devices. They simply ignore the offered ptime in SDP and stick to sending 30ms RTP packets no matter what. The last time I tested with a FRITZ!Box CPE was in February and I am absolutely sure that FS used to auto fix this type of problems in the past and when this was happening I would see the following lines in the logs: "Asynchronous PTIME not supported, changing our end from 20 to 30" "Changing Codec from G722@20ms@8000hz to G722@30ms@8000hz" and also something to do with timer resolution being increased to deal with 30ms intervals. This doesn't happen any more (v1.2.14) and all I can hear is choppy audio from these CPEs. The only relevant lines I could find in the logs were: "mod_conference.c:3590 Setup timer soft success interval: 20 samples: 160" "switch_core_io.c:759 Engaging Read Buffer at 640 bytes vs 240" I also verified with a packet trace that FS is indeed receiving RTP packets with payload of 240 bytes, which would correspond to 30ms and is sending packets with 160 bytes of payload (20ms), so as far as I can tell this would actually classify as async ptime. I only see the problem one way - the choppy audio comes from the remote party, which is sending the 30ms packets into the conference. The remote party however still hears normal audio coming from the conference (although it is packetized and sent out at 20ms). Could it be that for some reason FS or the conference application is mistakenly treating the 30ms packets as 20ms ones when mixing the audio into the conference? I do recognize that this is caused by misbehaving end device and it is not a FS issue, but I just wanted to know if someone else is also seeing similar problems or if this can somehow be related to a recent change in the FS code? I have updated to v1.2.15 so I can retest, but I will only get a chance to do it on Friday. Best Hristo
See if this article is helpful http://wiki.snom.com/FAQ/How_to_fix_my_Snom_phone's_audio_problems_with_Fritzbox%3F Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support@nysolutions.com<mailto:support@nysolutions.com> Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [cid:image001.jpg@01C72F94.9EE45D60]<http://www.nysolutions.com/> Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Hristo Trendev Sent: Monday, December 09, 2013 6:35 AM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] autofix incorrect packetization Hi, I get choppy audio in conferences from endpoints that send G722@30ms packets. The endpoint in question is a FRITZ!Box CPE and this seems to be a known problem with these type of devices. They simply ignore the offered ptime in SDP and stick to sending 30ms RTP packets no matter what. The last time I tested with a FRITZ!Box CPE was in February and I am absolutely sure that FS used to auto fix this type of problems in the past and when this was happening I would see the following lines in the logs: "Asynchronous PTIME not supported, changing our end from 20 to 30" "Changing Codec from G722@20ms@8000hz to G722@30ms@8000hz" and also something to do with timer resolution being increased to deal with 30ms intervals. This doesn't happen any more (v1.2.14) and all I can hear is choppy audio from these CPEs. The only relevant lines I could find in the logs were: "mod_conference.c:3590 Setup timer soft success interval: 20 samples: 160" "switch_core_io.c:759 Engaging Read Buffer at 640 bytes vs 240" I also verified with a packet trace that FS is indeed receiving RTP packets with payload of 240 bytes, which would correspond to 30ms and is sending packets with 160 bytes of payload (20ms), so as far as I can tell this would actually classify as async ptime. I only see the problem one way - the choppy audio comes from the remote party, which is sending the 30ms packets into the conference. The remote party however still hears normal audio coming from the conference (although it is packetized and sent out at 20ms). Could it be that for some reason FS or the conference application is mistakenly treating the 30ms packets as 20ms ones when mixing the audio into the conference? I do recognize that this is caused by misbehaving end device and it is not a FS issue, but I just wanted to know if someone else is also seeing similar problems or if this can somehow be related to a recent change in the FS code? I have updated to v1.2.15 so I can retest, but I will only get a chance to do it on Friday. Best Hristo
Hi Moishe,
See if this article is helpful http://wiki.snom.com/FAQ/How_to_fix_my_Snom_phone's_audio_problems_with_Fritzbox%3F
Thanks for pointing out. I'm also having trouble calling with codecs G722 to people with a Fritzbox. I'm hearing sound very well, but the b-leg hears my voice in slower motion. Because of that issue, I disabled the G722 codecs in my Baresip client. Annoying. Changing the software of all the Fritzboxs ourself is a bit too much to ask. That's actually the job of AVM (avm.de). -- EL
Hi,
See if this article is helpful http://wiki.snom.com/FAQ/How_to_fix_my_Snom_phone's_audio_problems_with_Fritzbox%3F
Thanks! I have actually seen this already, but that's not the point here, because I don't have control of the CPE. FS used to workaround this problem at the network side even with the broken Fritz!Box CPEs. I guess I am asking if this functionality was somehow lost in the newer releases and if somebody else with a recent FS version is also seeing this problem. BR, Hristo
It should still be there. You cold make sure you don't have rtp-autofix-timing set to false in your config. If you can't get it to work you could file a Jira on the issue, however I am not sure I have a way to verify it as I have no broken devices. On Mon, Dec 9, 2013 at 10:14 AM, Hristo Trendev <dist.lists@gmail.com>wrote:
Hi,
See if this article is helpful http://wiki.snom.com/FAQ/How_to_fix_my_Snom_phone's_audio_problems_with_Fritzbox%3F
Thanks! I have actually seen this already, but that's not the point here, because I don't have control of the CPE. FS used to workaround this problem at the network side even with the broken Fritz!Box CPEs. I guess I am asking if this functionality was somehow lost in the newer releases and if somebody else with a recent FS version is also seeing this problem.
BR, Hristo
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Hi, I tested again with the latest 1.2.16 release and the problem was still present. I'll make another test this week with 1.2.5.3. I am intentionally picking such an old version as I remember seeing this working at some point and I want to confirm that this is still the case. BTW, setting rtp-autofix-timing to either true or false made no difference. Best, Hristo On Mon, Dec 9, 2013 at 5:44 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
It should still be there. You cold make sure you don't have rtp-autofix-timing set to false in your config. If you can't get it to work you could file a Jira on the issue, however I am not sure I have a way to verify it as I have no broken devices.
On Mon, Dec 9, 2013 at 10:14 AM, Hristo Trendev <dist.lists@gmail.com>wrote:
Hi,
See if this article is helpful http://wiki.snom.com/FAQ/How_to_fix_my_Snom_phone's_audio_problems_with_Fritzbox%3F
Thanks! I have actually seen this already, but that's not the point here, because I don't have control of the CPE. FS used to workaround this problem at the network side even with the broken Fritz!Box CPEs. I guess I am asking if this functionality was somehow lost in the newer releases and if somebody else with a recent FS version is also seeing this problem.
BR, Hristo
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@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-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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@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-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
participants (4)
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Anthony Minessale
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EL
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Hristo Trendev
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Moishe Grunstein