- FreeSWITCH-users - FreeSWITCH Mailing Lists

Detect video presence in mod_conference
by Mateus Dalepiane 08 Apr '15

08 Apr '15

Hello, I am looking for a mechanism to detect whether the user that currently holds the conference video floor is actually sending a video stream or not. So far I know that the variable CF_VIDEO may be used to check if the channel is video capable, either to send and/or receive video. But this is not enough to ensure that a user is sending video, just if he is able to send and/or receive video. My idea is to check if there are video packets flowing through FS, that would mean that the user holding the floor is sending video. Is there a better way to do this? If not, where would be the best place to monitor the video packets flow? Thanks!

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Upgrade Freeswitch to v1.4 stable version
by Zvi Agmon 08 Apr '15

08 Apr '15

Hello, I'm trying to figure out what is the correct and best way to upgrade a server running Freeswitch v1.2 to v1.4 stable. I could not find any clear documentation - can you please help? Thanks a lot Zvi Agmon Lexifone <http://www.lexifone.com/> email: zvi(a)lexifone.com Office: +972-4-6817711 Cell: +972-54-4505109

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Re: [Freeswitch-users] WebRTC + Chrome = RTP Socket Error!
by Luiz Fernando Softov 07 Apr '15

07 Apr '15

*Brian West,* *I try to send with Re: but sometimes gmail remove Re: from subject* -- Luiz Fernando Softov http://www.softov.com.br fernando(a)softov.com.br

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Re: [Freeswitch-users] WebRTC + Chrome = RTP Socket Error!
by Luiz Fernando Softov 07 Apr '15

07 Apr '15

There is the Trace https://pastebin.freeswitch.org/24094 -- Luiz Fernando Softov http://www.softov.com.br fernando(a)softov.com.br

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vMobile configuration from Vitelity
by Sean Devoy 07 Apr '15

07 Apr '15

Does anyone have a vMobile device from Vitelity working through FS? I have outgoing from the cell phone working great, but the device never shows up in "show registrations" so I cannot route calls TO IT. Thanks, Sean

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Testing WebRTC / Issues with calls WebRTC/Softphones & Softphones/WebRTC
by Victor Medina 07 Apr '15

07 Apr '15

Hi! I am trying to test Freeswitch WebRTC capabiities. We are using JsSIP try web site (http://tryit.jssip.net/) and Zoiper Our servers are located on Amazon EC2 so, there is a NAT firewall between them. Freeswitch version is: FreeSWITCH Version 1.5.15b+git~20150404T093935Z~6eb59c7119~64bit (git 6eb59c7 2015-04-04 09:39:35Z 64bit) - TODAY MASTER. When testing WebRTC and Softphones, I find that I am able to call from a softphone to a WebRTC ext without problems. But when calling to from the WeRTC ext to the softphone it always fails with a Bad Media Description Can someone try to help in here? My relevant vars.xml lines: <X-PRE-PROCESS cmd="set" data="domain=ve001.voice.cibersys.com"/> <X-PRE-PROCESS cmd="set" data="domain_name=ve001.voice.cibersys.com"/> <X-PRE-PROCESS cmd="set" data="hold_music=local_stream://moh"/> <X-PRE-PROCESS cmd="set" data="use_profile=external"/> <X-PRE-PROCESS cmd="set" data="rtp_secure_media=optional:AES_CM_128_HMAC_SHA1_32"/> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=OPUS,PCMA,PCMU,G729,G722,H264,VP8,H263,H263-1998,H263-2000"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=OPUS,PCMA,PCMU,G729,G722,H264,VP8,H263,H263-1998,H263-2000"/> <X-PRE-PROCESS cmd="set" data="bind_server_ip=54.186.52.106"/> <X-PRE-PROCESS cmd="set" data="external_rtp_ip=54.186.52.106"/> <X-PRE-PROCESS cmd="set" data="external_sip_ip=54.186.52.106"/> <X-PRE-PROCESS cmd="set" data="sip_tls_version=tlsv1,tlsv1.1,tlsv1.2"/> <X-PRE-PROCESS cmd="set" data="sip_tls_ciphers=ALL:!ADH:!LOW:!EXP:!MD5:@STRENGTH"/>  <X-PRE-PROCESS cmd="set" data="internal_auth_calls=true"/> <X-PRE-PROCESS cmd="set" data="internal_sip_port=5090"/> <X-PRE-PROCESS cmd="set" data="internal_tls_port=5091"/> <X-PRE-PROCESS cmd="set" data="internal_ssl_enable=true"/> <X-PRE-PROCESS cmd="set" data="internal_ssl_dir=/opt/fs-20150407/certs/"/>  <X-PRE-PROCESS cmd="set" data="external_auth_calls=false"/> <X-PRE-PROCESS cmd="set" data="external_sip_port=5070"/> <X-PRE-PROCESS cmd="set" data="external_tls_port=5071"/> <X-PRE-PROCESS cmd="set" data="external_ssl_enable=false"/> My internal_profile <param name="liberal-dtmf" value="true"/> <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="aggressive-nat-detection" value="true"/> <param name="multiple-registrations" value="true"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="NDLB-received-in-nat-reg-contact" value="true"/> <param name="NDLB-force-rport" value="true"/> <param name="NDLB-broken-auth-hash" value="true"/> <param name="enable-timer" value="false"/> <param name="auth-calls" value="true"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="wss-binding" value=":7443"/> Thanks you as usual for your kind help. -- Víctor E. Medina M. Platform Architect / Chief Infrastructure +58424 291 4561 BB #79A8AFA2 @VMCibersys

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Testing WebRTC / Issues with Video/Audio calls to ext echotest/9196 and Conference
by Victor Medina 07 Apr '15

07 Apr '15

Hi! I am trying to test Freeswitch WebRTC capabiities. We are using JsSIP try web site (http://tryit.jssip.net/) Our servers are located on Amazon EC2 so, there is a NAT firewall between them. Freeswitch version is: FreeSWITCH Version 1.5.15b+git~20150404T093935Z~6eb59c7119~64bit (git 6eb59c7 2015-04-04 09:39:35Z 64bit) - TODAY MASTER. When calling to echoetst ext. 9196, and using the "video/audio" call it works perfectly, when calling without the Video enable call is placed but no audio is heard. The same thing happens when calling to the ext 3000 ext conference. When using the video enabled everything seems to work just fine. But if I disable video tha call wont work. Can someone try to help in here? My relevant vars.xml lines: <X-PRE-PROCESS cmd="set" data="domain=ve001.voice.cibersys.com"/> <X-PRE-PROCESS cmd="set" data="domain_name=ve001.voice.cibersys.com"/> <X-PRE-PROCESS cmd="set" data="hold_music=local_stream://moh"/> <X-PRE-PROCESS cmd="set" data="use_profile=external"/> <X-PRE-PROCESS cmd="set" data="rtp_secure_media=optional:AES_CM_128_HMAC_SHA1_32"/> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=OPUS,PCMA,PCMU,G729,G722,H264,VP8,H263,H263-1998,H263-2000"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=OPUS,PCMA,PCMU,G729,G722,H264,VP8,H263,H263-1998,H263-2000"/> <X-PRE-PROCESS cmd="set" data="bind_server_ip=54.186.52.106"/> <X-PRE-PROCESS cmd="set" data="external_rtp_ip=54.186.52.106"/> <X-PRE-PROCESS cmd="set" data="external_sip_ip=54.186.52.106"/> <X-PRE-PROCESS cmd="set" data="sip_tls_version=tlsv1,tlsv1.1,tlsv1.2"/> <X-PRE-PROCESS cmd="set" data="sip_tls_ciphers=ALL:!ADH:!LOW:!EXP:!MD5:@STRENGTH"/>  <X-PRE-PROCESS cmd="set" data="internal_auth_calls=true"/> <X-PRE-PROCESS cmd="set" data="internal_sip_port=5090"/> <X-PRE-PROCESS cmd="set" data="internal_tls_port=5091"/> <X-PRE-PROCESS cmd="set" data="internal_ssl_enable=true"/> <X-PRE-PROCESS cmd="set" data="internal_ssl_dir=/opt/fs-20150407/certs/"/>  <X-PRE-PROCESS cmd="set" data="external_auth_calls=false"/> <X-PRE-PROCESS cmd="set" data="external_sip_port=5070"/> <X-PRE-PROCESS cmd="set" data="external_tls_port=5071"/> <X-PRE-PROCESS cmd="set" data="external_ssl_enable=false"/> My internal_profile <param name="liberal-dtmf" value="true"/> <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="aggressive-nat-detection" value="true"/> <param name="multiple-registrations" value="true"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="NDLB-received-in-nat-reg-contact" value="true"/> <param name="NDLB-force-rport" value="true"/> <param name="NDLB-broken-auth-hash" value="true"/> <param name="enable-timer" value="false"/> <param name="auth-calls" value="true"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="wss-binding" value=":7443"/> Here is the log of the call to 9196 without audio and video (no video option selected) http://pastebin.com/AncwpiPx <- this doesn't work Here is the log of the call to 9196 with audio and video option selected http://pastebin.com/FT7XMpzU <- in this one audio and video works -- Víctor E. Medina M. Platform Architect / Chief Infrastructure +58424 291 4561 BB #79A8AFA2 @VMCibersys

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DOC Sprint Apr 10
by Ítalo Rossi 07 Apr '15

07 Apr '15

Hi Everyone, We are organizing an other Sprint to work on FreeSWITCH Documentation on Friday 10 April 2015 at 10am CT. It will be 4 hours long but you can join for less time. The Doc-sprint will focus on migrating the remaining pages from old FS Wiki (https://wiki.freeswitch.org) to Confluence ( https://freeswitch.org/confluence) We will use this IRC channel during the sprint: #freeswitch-docs and we will be tracking our work on a spreadsheet: https://docs.google.com/spreadsheets/d/1qsG-kRymvKlNBapnBLw86W130VdbnK6naYa… to avoid working on the same content. So during the sprint, please change the page ""Status"" you are working on to ""Editing"" with your name next to it. Some extra information: - https://freeswitch.org/confluence/display/FREESWITCH/Wiki+Migration - https://freeswitch.org/confluence/display/FREESWITCH/Contributing+Documenta… Let us know if you have any question. We hope to get a maximum number of people signed up! So, who is in? Confirmed: Italo (+4) Areski Bote -- Ítalo Rossi

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WebRTC + Chrome = RTP Socket Error!
by Luiz Fernando Softov 07 Apr '15

07 Apr '15

I'm testing WebRTC + Chrome Using http://sipjs.com When im using FireFox, its work fine, i can make and receive calls with audio and/or video But when i try to use chrome. The other extension rings, but when i answer the call I receive this error, and call hangup 2015-04-07 05:51:39.016628 [ERR] switch_core_media.c:5775 AUDIO RTP REPORTS ERROR: [Socket Error!] 2015-04-07 05:51:39.016628 [NOTICE] switch_core_media.c:5776 Hangup sofia/internal/1001(a)192.168.1.100 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] Has anyone experienced the same problem? -- Luiz Fernando Softov http://www.softov.com.br fernando(a)softov.com.br

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Originate_timeout longer than 5 minutes?
by Fredrik Larsson 07 Apr '15

07 Apr '15

Hi, I am sending originate_timeout=600 on ESL but it always terminates after 5 minutes e.g. 300 seconds. I have tried to find out if there is some hardcoded max variable but I can't find it. Do anyone know if there is some other variable I can change to get higher value or is this per design? Regards, Fredrik

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