April 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Freeswitch and Ping
by Clive Lansink 22 Apr '16

22 Apr '16

I did wonder about that but then I tried it on the LAN and got the same results. Clive Lansink Email: Clive(a)Lansink.Co.NZ Phone: +64 9 520-4242 Mobile: +64 21 663-999 Fax: +64 21 789-150 -----Original message----- From: Michael Jerris <mike(a)jerris.com> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Subject: Re: [Freeswitch-users] Freeswitch and Ping Reply-to: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Date: Wed, 20 Apr 2016 22:52:30 -0400 sounds like a broken router to me On Wednesday, April 20, 2016, Clive Lansink <clive(a)lansink.co.nz> wrote: > > Hi everyone. I often use Freeswitch set up as a client to provide a soft > phone on my laptop. > > But recently I noticed a significant problem when at the same time I was > running a standard Windows "ping" in the background. The speech I heard > from the other end would often stutter, and the remote party complained > that my speech sometimes sounded gittery. > > I reasoned that ping doesn't take up much network bandwidth so while in > conversation I did something like copy a large file from my laptop to > another PC on the network and there was no disturbance to the conversation. > It only seems to be ping that causes this. > > Is there something about pinging that might impact on Freeswitch or VOIP? > What would cause this? > > Further information. > > the above refers to the standard Windows ping command. I also have a > Python script that pings a specified host, and running it also caused the > call quality to degrade. I tried various changes to the source code to no > effect, including: > * only creating the socket once and then reusing it instead of it being > created for each ping. > * Only creating the data packet once and reusing it instead of it being > created for each ping. > * Using a very short packet size. > * Eliminating the select statement and just sleeping for a period of time > before reading the response in case the select itself was at fault. > * Eliminating the step of dissecting and decoding the received packet. > None of these made any difference so it seems to me that whatever it is > must be something quite fundamental to something at the lower networking > level. > > Thank you. > > > Clive Lansink > Email: Clive(a)Lansink.Co.NZ <javascript:;> > Phone: +64 9 520-4242 > Mobile: +64 21 663-999 > Fax: +64 21 789-150 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org <javascript:;> > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org <javascript:;> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.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 http://www.freeswitch.org

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Verto latency
by Gregor Nanger 22 Apr '16

22 Apr '16

If I make call from SIP client everything is ok. But if I make call from verto demo client, there is 2 second delay of audio in direction from browser (latest chrome) to callee. Other direction is ok. Is it posible to tweak lower this latency?

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Nat issue
by Miha 22 Apr '16

22 Apr '16

Hi our FS is on public ip but the voip server to which trunk is made (SFB) is behind NAT. So the issue is that in SDP is private ip and FS sends RTP to this private ip, it does not change to src ip from which sip request was received. Remote SDP: 2016-04-21 08:04:00.954752 [DEBUG] sofia.c:5614 Remote SDP: v=0 o=- 46176 1 IN IP4 172.16.2.3 s=session c=IN IP4 172.16.2.3 b=CT:1000 t=0 0 m=audio 31200 RTP/AVP 8 101 13 c=IN IP4 172.16.2.3 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=rtcp:31201 a=label:Audio And then: xxx.xxx.xxx.xxx (FS public ip) port 17094 -> 172.16.2.3 port 31200 codec: 8 ms: 30 I tried with: <param name="NDLB-force-rport" value="true"/> <param name="aggressive-nat-detection" value="true"/> on internal profile and on trunk profile but the same. tnx for help! Miha

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Codec negotiation issue
by Mickael Hubert 22 Apr '16

22 Apr '16

Hi list, I have an issue with the negociation codec in my Freeswitch. I followed this doc: https://wiki.freeswitch.org/wiki/Codec_Negotiation The call flow: 1) INVITE: UAC -- (G729, PCMA) --> FS -- (G729, PCMA) --> SVI Asterisk 2) 200OK UAC <-- (G729) -- FS <-- (PCMA) -- SVI Asterisk (HOMER) I have inherit_codec=true in my dialplan and inbound-late-negotiation true in sip-profile. But freeswitch do not force codec learned from leg B to leg A. *LEG B (200OK):* *2016-04-01 11:28:16.453784 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [PCMA:8:8000:150:64000]/[G729:18:8000:20:8000]2016-04-01 11:28:16.453784 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [PCMA:8:8000:150:64000]/[PCMA:8:8000:20:64000]2016-04-01 11:28:16.453784 [DEBUG] switch_core_media.c:3248 Audio Codec Compare [PCMA:8:8000:20:64000] ++++ is saved as a match2016-04-01 11:28:16.453784 [DEBUG] switch_core_codec.c:111 sofia/internal/06********@HOMER Original read codec set to PCMA:8* LEG B is in G711A, OK, next: 2016-04-01 11:28:16.453784 [NOTICE] sofia.c:6727 Channel [sofia/internal/06*******@HOMER] has been answered 2016-04-01 11:28:16.453784 [DEBUG] switch_channel.c:3686 (sofia/internal/06*********@HOMER) Callstate Change DOWN -> ACTIVE 2016-04-01 11:28:16.473813 [DEBUG] switch_ivr_originate.c:412 *Setting codec string on sofia/external/0326793005(a)1.1.1.1 <0326793005(a)1.1.1.1> to PCMA@8000h@20i* We can see, FS set the correct codec to LEG A; good ! , next... *2016-04-01 11:28:16.473813 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [G729:18:8000:20:8000]/[G729:18:8000:20:8000]2016-04-01 11:28:16.473813 [DEBUG] switch_core_media.c:3248 Audio Codec Compare [G729:18:8000:20:8000] ++++ is saved as a match2016-04-01 11:28:16.473813 [DEBUG] switch_core_media.c:3194 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000]2016-04-01 11:28:16.473813 [DEBUG] switch_core_media.c:3248 Audio Codec Compare [PCMA:8:8000:20:64000] ++++ is saved as a match2016-04-01 11:28:16.473813 [DEBUG] switch_core_media.c:2139 Set Codec sofia/external/0326793005(a)1.1.1.1 <0326793005(a)1.1.1.1> G729/8000 20 ms 160 samples 8000 bits* Why FS compare and re set a new codec (G729) ? Next: *2016-04-01 11:28:16.613776 [ERR] mod_g729.c:145 This codec is only usable in passthrough mode!2016-04-01 11:28:16.613776 [ERR] switch_core_io.c:1245 Codec G.729 decoder error!* normal .... *For informations:* - FreeSWITCH Version 1.5.8b+git~20140214T000311Z~fe2a4d6d47~64bit (git fe2a4d6 2014-02-14 00:03:11Z 64bit) Is it a bug ? an configuration error ? Thanks in advance PS: *My dialplan:* *<condition field="${destination_number}" expression="^0([1-79][0-9]{8})$" break="on-true"><action application="export" data="codec_string=${ep_codec_string}"/><action application="set" data="inherit_codec=true"/><action application="bridge" data="sofia/internal/${destination_number}@HOMER"/></condition>* -- Cordialement HUBERT Mickaël Ingénieur VOIP - Hexanet --

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London Marathon 2016
by Andrew Cassidy 22 Apr '16

22 Apr '16

Hi Guys, Not like me to spam this sort of stuff to the list, but it's worth a go! Sunday is the London Marathon. My other half is running this year for the charity WhizzKidz. They're a UK based charity who supply equipment to disabled children, more information can be found here: http://www.whizz-kidz.org.uk/ Any contributions would be much appreciated. Here's the link to her justgiving page: https://www.justgiving.com/Rachel-Wilson39/ Thanks for your time! Kind regards, -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director 03303 880 960 andrew(a)cassidyweb.co.uk <andrew(a)cassidywebservices.co.uk> www.cassidyweb.co.uk <http://www.cassidywebservices.co.uk>

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How to send Re_invite
by amani mansour 22 Apr '16

22 Apr '16

Hi all, I need to configure FS to send 491 request pending message after receiving Re invite from the caller , The problem that i don't know how to send the re invite request ,which xml file i whould modifie it ? please help me ,i have done many research but i didn't find anything . Thank you , Best Regards Amani

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C11 style in libyuv?
by Luis Daniel Lucio Quiroz 21 Apr '16

21 Apr '16

Hello Is it my imagination of I see c11 style in libyuv (the one bundled in fs 1.6.7)? This may discard older compilers. Ld

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mod_http_cache doesn't retrieve any file
by Alexander Mustafin 21 Apr '16

21 Apr '16

Hi there! I am testing mod_http_cache module and unfortunately it doesn’t work for me properly. I loaded module, and trying console commands such as ‘http_prefetch’ and others. A respond of server look like: freeswitch(a)fs.sip3.net> http_prefetch http://download.wavetlan.com/SVV/Media/HTTP/WAV/Media-Convert/Media-Convert… +OK 2016-04-20 12:22:38.810363 [DEBUG] mod_http_cache.c:565 Locked cache 2016-04-20 12:22:38.810363 [INFO] mod_http_cache.c:653 Cache MISS: size = 1 (0 MB), hit ratio = 0/2 2016-04-20 12:22:38.810363 [DEBUG] mod_http_cache.c:718 Adding http://download.wavetlan.com/SVV/Media/HTTP/WAV/Media-Convert/Media-Convert… to cache index 1 2016-04-20 12:22:38.810363 [DEBUG] mod_http_cache.c:576 Unlocked cache 2016-04-20 12:22:38.810363 [DEBUG] mod_http_cache.c:1069 opening /var/cache/freeswitch/c9/396ca9-b29f-4a97-b40d-cb1430a7f1b6.wav for URL cache 2016-04-20 12:22:38.810363 [ERR] mod_http_cache.c:1103 open() error: No such file or directory 2016-04-20 12:22:38.810363 [DEBUG] mod_http_cache.c:565 Locked cache 2016-04-20 12:22:38.810363 [INFO] mod_http_cache.c:675 Failed to download URL http://download.wavetlan.com/SVV/Media/HTTP/WAV/Media-Convert/Media-Convert… 2016-04-20 12:22:38.810363 [DEBUG] mod_http_cache.c:576 Unlocked cache And server does not try to obtain file from the url, because I listen 80 port and I can’t see anything. Possibly the problem here '2016-04-20 12:22:38.810363 [INFO] mod_http_cache.c:653 Cache MISS: size = 1 (0 MB), hit ratio = 0/2’ but I don’t know what it’s about. Any clue for this problem? Best regards, Alexander Mustafin mustafin.aleksandr(a)gmail.com

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Re: [Freeswitch-users] Freeswitch and Ping
by Michael Jerris 21 Apr '16

21 Apr '16

sounds like a broken router to me On Wednesday, April 20, 2016, Clive Lansink <clive(a)lansink.co.nz> wrote: > > Hi everyone. I often use Freeswitch set up as a client to provide a soft > phone on my laptop. > > But recently I noticed a significant problem when at the same time I was > running a standard Windows "ping" in the background. The speech I heard > from the other end would often stutter, and the remote party complained > that my speech sometimes sounded gittery. > > I reasoned that ping doesn't take up much network bandwidth so while in > conversation I did something like copy a large file from my laptop to > another PC on the network and there was no disturbance to the conversation. > It only seems to be ping that causes this. > > Is there something about pinging that might impact on Freeswitch or VOIP? > What would cause this? > > Further information. > > the above refers to the standard Windows ping command. I also have a > Python script that pings a specified host, and running it also caused the > call quality to degrade. I tried various changes to the source code to no > effect, including: > * only creating the socket once and then reusing it instead of it being > created for each ping. > * Only creating the data packet once and reusing it instead of it being > created for each ping. > * Using a very short packet size. > * Eliminating the select statement and just sleeping for a period of time > before reading the response in case the select itself was at fault. > * Eliminating the step of dissecting and decoding the received packet. > None of these made any difference so it seems to me that whatever it is > must be something quite fundamental to something at the lower networking > level. > > Thank you. > > > Clive Lansink > Email: Clive(a)Lansink.Co.NZ <javascript:;> > Phone: +64 9 520-4242 > Mobile: +64 21 663-999 > Fax: +64 21 789-150 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org <javascript:;> > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org <javascript:;> > 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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Freeswitch and Ping
by Clive Lansink 21 Apr '16

21 Apr '16

Hi everyone. I often use Freeswitch set up as a client to provide a soft phone on my laptop. But recently I noticed a significant problem when at the same time I was running a standard Windows "ping" in the background. The speech I heard from the other end would often stutter, and the remote party complained that my speech sometimes sounded gittery. I reasoned that ping doesn't take up much network bandwidth so while in conversation I did something like copy a large file from my laptop to another PC on the network and there was no disturbance to the conversation. It only seems to be ping that causes this. Is there something about pinging that might impact on Freeswitch or VOIP? What would cause this? Further information. the above refers to the standard Windows ping command. I also have a Python script that pings a specified host, and running it also caused the call quality to degrade. I tried various changes to the source code to no effect, including: * only creating the socket once and then reusing it instead of it being created for each ping. * Only creating the data packet once and reusing it instead of it being created for each ping. * Using a very short packet size. * Eliminating the select statement and just sleeping for a period of time before reading the response in case the select itself was at fault. * Eliminating the step of dissecting and decoding the received packet. None of these made any difference so it seems to me that whatever it is must be something quite fundamental to something at the lower networking level. Thank you. Clive Lansink Email: Clive(a)Lansink.Co.NZ Phone: +64 9 520-4242 Mobile: +64 21 663-999 Fax: +64 21 789-150

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