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RTP ptime changed multiple times
by Jose Fco. Irles Durá 19 Sep '22

19 Sep '22

Hello, I have a problem with some calls where the rtp stream from the other side changes the rtp ptime multiple times. I make a call with the sdp line "a=ptime:20" but the initial rtp comes with 40ms ptime. FreeSWITCH detects it and changes the ptime to 40ms and all works ok. Later in the conversation, without any re-invite, the rtp stream changed to 20ms. At this moment, FreeSWITCH does not detect the change and the audio sended by FreeSWITCH to the other legs is choppy. I think that the problem is on the other side, but the other side is a big operator and always this is a problem to solve issues. I tried to set "rtp_media_autofix_timing", but if I set this property, the first part of the conversation is discarded (ptime at 40ms, silence in my phone) and later I can listen the audio (when the ptime=20ms). Is there any solution for this? Best regards -- Jose Fco. Irles Durá

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Re: [Freeswitch-users] RTP ptime changed multiple times
by Sam 19 Sep '22

19 Sep '22

Hello Jose Sorry to interrupt the thread, but just wondering if you think adding a maxptime attribute in the SDP may be a work around for you? If you set this to 20, the switch between 20/40 should not happen. Kind regards Sameee From: Jose Fco. Irles Dur <josefu(a)gmail.com> Sent: Mon, 19 Sep 2022 11:59:42 GMT+0200 To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Subject: Re: [Freeswitch-users] RTP ptime changed multiple times iWe couldn't verify the sender of this email. Tell the other side to fix their brokenness maybe?  How did you try setting the flag? Yes, I know that the problem is from the provider, but its response time is bad. I setted "rtp_media_autofix_timing=false" for testing, but when I setted, the first packets (40ms) are ignored and when the ptime is 20ms I listen to it. I think this is the correct behavior according to my understanding, so I can't solve the problem with the buggy provider. Best regards  /b On Fri, Sep 16, 2022 at 4:28 PM Jose Fco. Irles Durá <josefu(a)gmail.com> wrote: Hello, I have a problem with some calls where the rtp stream from the other side changes the rtp ptime multiple times. I make a call with the sdp line "a=ptime:20" but the initial rtp comes with 40ms ptime.FreeSWITCH detects it and changes the ptime to 40ms and all works ok.Later in the conversation, without any re-invite, the rtp stream changed to 20ms. At this moment, FreeSWITCH does not detect the change and the audio sended by FreeSWITCH to the other legs is choppy. I think that the problem is on the other side, but the other side is a big operator and always this is a problem to solve issues. I tried to set "rtp_media_autofix_timing", but if I set this property, the first part of the conversation is discarded (ptime at 40ms, silence in my phone) and later I can listen the audio (when the ptime=20ms). Is there any solution for this? Best regards -- Jose Fco. Irles Durá _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community Professional FreeSWITCH Services sales(a)freeswitch.com https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com -- Brian West | Co-founder and DeveloperNeed Commercial support? email sales(a)freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045Email: brian(a)freeswitch.comMobile: 918-424-9378Website: https://www.FreeSWITCH.com _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community Professional FreeSWITCH Services sales(a)freeswitch.com https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com -- Jose Fco. Irles Durá _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community Professional FreeSWITCH Services sales(a)freeswitch.com https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com

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after failed Call: Failed Registration with status Request Timeout [408].
by Philipp Zeitschel 15 Sep '22

15 Sep '22

Hi, i've got a newly setup system with fusionPBX. I have configured my trunk and it registers successfully and is useable. I can send outgoing calls and so on. But if i receive an incoming call that has no destination inside of the system, the trunk disconnects after a few seconds, and does not manage to reregister again, i get only this messages: 2022-09-15 07:14:11.316316 98.00% [ERR] sofia_reg.c:2677 d8d092e4-2f22-4dda-95a6-4e85dd4e2862 Failed Registration with status Request Timeout [408]. failure #1 2022-09-15 07:14:12.016310 98.00% [WARNING] sofia_reg.c:516 d8d092e4-2f22-4dda-95a6-4e85dd4e2862 Failed Registration [408], setting retry to 30 seconds. 2022-09-15 07:14:43.116326 98.27% [NOTICE] sofia_reg.c:459 Registering d8d092e4-2f22-4dda-95a6-4e85dd4e2862 Only after a reboot it comes back online. This sounds exactly as my problem: https://freeswitch-users.freeswitch.narkive.com/vsrnW2f4/gateway-registratio n-issue-can-only-be-solved-through-killgw is this a known issue? Regards Philipp

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Correct syntax for sending out an SMS message trough external SIP provider
by Jelle de Jong 14 Sep '22

14 Sep '22

Hello everybody, I am trying to use chat sip to sent an SMS message but I can not get the command to work. # fs_cli -x "sofia status" | grep REGED external::speakup-powercraft gateway sip:powercraft@alg.sip.speakup.nl REGED If I would call the number it would look like this: sofia_glue.c:1654 sofia/external/316xxxxxxxx sending invite version: 1.10.7 -release-19-883d2cb662 64bit fs_cli -x "chat sip|noreply@mydomain|external/sip:+316xxxxxxxx@speakup-gateway|Hello" What would be the syntax to sent out a SMS from 3161218xxxx to 3161234xxxx (both numbers are registered at the external gateway). Is this possible with the above or am I mislead, can someone give me an example? https://freeswitch.org/confluence/display/FREESWITCH/mod_sms https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+chat Kind regards, Jelle de Jong

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nginx and FreeSWITCH 1.10
by Kakiman Expert 08 Sep '22

08 Sep '22

hello does anybody has experience using nging as wss to ws proxy ? (usage of nginx to decrypt wss and then forward ws to FreeSWITCH) thanks franck

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Fwd: websocket disconnection issue - error 1006
by Kakiman Expert 06 Sep '22

06 Sep '22

Hello I am working on FreeSWITCH 1.10.6 ans sip.js 0.16, and very often, I have a deconnection of the WSS session. The error message on javascript console on Chrome is "error 1006 - websocket unexpectly closed" This issue is very random but very often, and it breaks the incoming call. Do you experiment such issue ? Do you know how to solve it ? thanks Franck

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FreeSWITCH WebSockets and SIP.js
by Jerry Kendall 06 Sep '22

06 Sep '22

Hey there... Been using FS for a few years now and it's been great. Thanks to the guys that maintain it.. I've added certs and set wss-binding as required and even create a great little WebPhone based on SIP.js Everything is fine except.... I keep getting 'websocket closed unexpectedly error 1006' I hit refresh and everything is fine again, for a while. Has anyone else used the websockets and hit this issue? If so, did you figure it out? There are pages on pages of complaints but can seem to find a fix. Jerry

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Too long Reason header swallowed
by Gilles SAUVAIRE 02 Sep '22

02 Sep '22

I often have this case. https://github.com/signalwire/freeswitch/issues/1481 does anyone find a workaround ? has anyone made a sofia patch ? This problem happens frequently for me. (Ms Teams send very long text cause) Thanks. Gilles

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VP8 conference FIR Signal not always sent when frames are dropped
by Josh H 26 Aug '22

26 Aug '22

Hi, I am using Verto to connect to FreeSwitch over webRTC in a conference call. This works fine most of the time however when using an older device that does not support hardware acceleration for the video encoding, it appears that some frames are dropped. The CPU is pegged at 100% utilisation, with 60-80% of that being from the chrome process. This is understandable when using older hardware however when this occurs FreeSwitch fails to send FIR packets to get a new keyframe, leading to the ‘video mute’ image being shown until the next scheduled keyframe (which with VP8 is, by default, around 3000 frames, equivalent to 100 seconds at 30fps). The client seems to be unaware of any issues as it continues to encode frames and send them over the network, leading me to believe it is an issue with FreeSwitch. My current solution is to enable the ‘kf-max-dist’ parameter in ‘vpx.conf.xml’ however this leads to occasional stuttering, I assume this is because it must encode a full frame every 10 seconds. The other solution I have tested is swapping to the H.264 codec (the only other codec supported by webRTC) which makes the issue less noticeable however the quality of the stream drops significantly when no FIR is requested. H.264 isn’t an option for me as some of my target clients are Android devices which do not support H.264 over webRTC. Given that sometimes the session works as expected with FIR packets being sent when needed, leading to a small video stutter before it recovers, I am unsure how to fix it or what logs etc to provide. Any pointers would be appreciated 😊 Thanks Josh

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Websocket closed unexpectly
by Kakiman Expert 24 Aug '22

24 Aug '22

Hello I am working on FreeSWITCH 1.10.6 ans sip.js 0.16, and very often, I have a deconnection of the WSS session. The error message on javascript console on Chrome is "error 1006 - websocket unexpectly closed" This issue is very random but very often, and it breaks the incoming call. Do you experiment such issue ? Do you know how to solve it ? thanks Franck

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