July 2018 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Stumped by mod_portaudio permissions issue
by Mark Haun 02 Feb '20

02 Feb '20

I built freeswitch version 1.7.0+git~20150730T192909Z~ab7f83c654~32bit for my ARM-based home server (an NVIDIA Jetson board), with the intention of using mod_portaudio with a wireless USB headset for my main home phone. It works well when running freeswitch directly as root, but when I configured freeswitch to run at system startup, mod_portaudio could no longer find any audio devices. No problem, I thought, it's a simple permissions issue. I am using the example debian init script from the wiki (this is a Jessie install with SysV init rather than systemd). It runs freeswitch as user freeswitch, group daemon using the -u and -g command-line options. So I added "freeswitch" to the audio group, then used sudo to verify that I could play sound files with "aplay" as the freeswitch user. That worked, but it did not solve the mod_portaudio failure. Thinking there must be an issue with portaudio separate from ALSA, I fetched and built the "pa_devs" utility which enumerates and prints the available audio devices. (This is basically what mod_portaudio is doing on startup.) Running as the freeswitch user, that worked too. Hmmmm. I tried running freeswitch from the command line using sudo rather than the -u and -g options, i.e. $ sudo freeswitch /usr/local/freeswitch/bin/freeswitch -nc and that worked as well! Then I started experimenting with -u and -g and discovered that $ /usr/local/freeswitch/bin/freeswitch -nc -u freeswitch allows mod_portaudio to load correctly, whereas $ /usr/local/freeswitch/bin/freeswitch -nc -u freeswitch -g daemon causes mod_portaudio to fail. Can anyone explain what's happening here? In both cases the process UID, GID, EUID, EGID, FUID, and FGID are the same (freeswitch:daemon) as displayed with "ps xao pid,uid,gid,euid,egid,fuid,fgid". Could this be a bug in freeswitch? There are a couple of threads from 2-3 years ago in the archives, where mod_portaudio permissions issues were never resolved and the solution remained a mystery. For the time being I guess this (not using -g) is a workaround, but I'd sure like to understand what is going on. Mark

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ICE/DTLS handshake
by Mirko Brankovic 25 Oct '19

25 Oct '19

HI, Has anyone experienced DTLS handshake takes 5s to get to SETUP state: > 2017-09-21 11:40:40.021178 [INFO] switch_rtp.c:3515 Changing audio DTLS > state from OFF to HANDSHAKE > 2017-09-21 11:40:45.319457 [INFO] switch_rtp.c:3172 Changing audio DTLS > state from HANDSHAKE to SETUP In network dump i see that answering side is not sending STUN for this 5s and then suddenly answers last 5 STUNs from A side. Has anyone encountered this kind of problem ? I have a pcap if necessary... Thanks, -- Regards, Mirko

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Change SIP Contact header on b-leg
by Кирилл Преслер 22 Feb '19

22 Feb '19

Provider not accepting calls because of "bad" Contact header sip:gw+ss@10.50.244.15:5060;transport=udp;gw=ss ITSP needs sip:username@10.50.244.15:5060, where username should be passed from dialplan for every call and there is no params. I've tried following variables with set and export, but nothing seems to work. sip_contact_user sip_outgoing_contact_uri sip_invite_contact_params

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DTMF negotiation
by kaiduan xie 04 Feb '19

04 Feb '19

Hi, We are encountering the following DTMF issue, FreeSwitch adds telephone-event support in INVITE to SBC, SBC replies back without telephone-event support in 200. However FreeSwitch still sends out DTMF in RFC 2833 event. I think FreeSwitch should send out DTMF inband instead of 2833 event, what is the configuration for this? Many thanks for help, /Kaiduan

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Sofia stops responding after a few days
by Bipin Patel 01 Feb '19

01 Feb '19

hi, I have 2 instances of FS running on a single windows box, first instance uses normal sip UDP profiles mainly used for routing calls to carriers and its running as a service since a few months without any issues. The second instance runs a sip TLS profile and accepts inbound registrations and forwards calls in sip UDP ahead, it uses xml_curl for directory users but the problem is every 2-3 days the inbound TLS profile stops responding to registrations, and when that happens i cant even stop and restart the service, have to kill it and start again. FS_CLI works but doesnt show any error, at first i though it could be the xml_curl causing the issue but later realized it never sends any requests when sofia stops responding. i have been banging my head from the past week or so but not able to find the cause, could any1 help in guiding me what to check when this happens so can find the root cause, im using commercial certs for TLS and when its running there r no issues other than sofia stops responding every few days, it happens at random times. -- Regards, Bipin ------------------------------------------------------------------------

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Audio delayed for about 3sec.
by 王聡 20 Dec '18

20 Dec '18

Hey all, Today I moved my freeswitch server from VM to cloud server, but I found there has an audio delay on both caller and receiver for about 3sec in each call during call test. Is there any idea for this audio delay? Regards.

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Re: [Freeswitch-users] h264 video call between endpoints with one end at bad resolution of 352x288
by Michael Jerris 23 Oct '18

23 Oct '18

sounds like you might be doing the option to mirror resolution > On Jun 28, 2018, at 9:07 AM, antonio via FreeSWITCH-users <freeswitch-users(a)lists.freeswitch.org> wrote: > > > From: antonio <asilva(a)wirelessmundi.com> > Subject: h264 video call between endpoints with one end at bad resolution of 352x288 > Date: June 26, 2018 at 6:05:20 AM EDT > To: freeswitch-users(a)lists.freeswitch.org > > > Hi, > > Calling between two Grandstream GXV3275 i expected to have a video call at 1280x720, but the video that is present at destination as the resolution of 352x288 and the image as poor quality, originator is sending video at 1280×720. If i swap the devices the result is the same, the destination always get a bad resolution of 352x288 but the originator receives the video at expected resolution of 1280x720. > > If i set rtp_direct between the two endpoints it work as expected. > > I also tested a conference room and force the it resolution to 1280*x720 it work well, if i don't force the canvas-size i see: > > 2018-06-26 13:55:02.077824 [WARNING] mod_conference.c:3157 Unspecified video-canvas-size, falling back to 1280x720 > > And the conference is done at 1280x720. > > > I've open a jira where you can find a full log with this behavior: > > https://freeswitch.org/jira/browse/FS-11140 > > > I don't understand why there is transcoding and mod_av is used, the codec negotiated is the same.. i understand that the resolution is at 352x288 for the first frames sent by the phone but then it send them at 1280x720, mod_av should detected and adjust the new resolution, no? > > > Anyone experience this behavior? >

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Freeswitch failed to initiate outbound call using SIPs + SRTP (SRTP unprotect )
by Chhorm Chhatra 22 Oct '18

22 Oct '18

Hello, Currently, I faced a problem regarding SRTP outbound call to user (Leg B). The scenario is like this, - We set up our own root CA to an IP address (e.g 192.168.0.13) - We create a server certificate for freeswitch at 192.168.0.13 - Linphone is used as SIP client and is configured to trust our root CA by default. - Linphone A is configured to register to Freeswitch vis TLS + SRTP. (One leg call to server has both SIPs and SRTP – completely secure) - Linphone B is registered to Freeswitch via TLS + SRTP, and waiting for Linphone A to call to. (One leg call to server, e.g. 9196 (echo test), is completely secure with SRTP + SIPs) - Unfortunately, if A call to B, only A leg has SIPs + SRTP, but Leg B is not encrypted with SRTP and SIPs at all. This causes *SRTP unprotect failed with code 7 (auth check failed)**.* + Dialplan Configuration <action application="set" data="rtp_secure_media=true"/> <action application="export" data="rtp_secure_media=true"/> The dial-string is <action application="bridge" data="user/${dialed_extension}@${domain_name}"/> + Directory Configruation: <param name="dial-string" value="{rtp_secure_media=${ regex(${sofia_contact(${dialed_user}@${dialed_domain}) }|transport=tls)},presence_id=${dialed_user}@${dialed_ domain}}${sofia_contact(${dialed_user}@${dialed_domain})}" /> My question is that, is there any configuration left that I have to set up in order to let freeswitch initiate an outbound call to Leg B correctly with SRTP and SIPs (tls)? Any help would be really appreciated. Thank you so much. Best Regard,

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Multiple concurrent Verto calls in browser
by Michael Avers 05 Oct '18

05 Oct '18

Hello, I'm using Verto Communicator as a base for building a simple app to make/receive calls - and it works great. I'm trying to add the ability to manage multiple concurrent calls. My first thought is to create a new Verto service (in Angular) that maintains call state and reference to currently active call and otherwise keeps all calls in an array. Then to switch calls it'd place the currently active one on hold and then unhold any of the other calls in the array. Does this make sense? Any tips on handling multiple calls with Verto? Do I need a separate websocket connection for each? (they're all using the same proxy and credentials). Thanks! Mike

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Tagged v.1.8.1 - will there be pre-built packages, Debian 9/CentOS 7?
by Richard Chan 04 Sep '18

04 Sep '18

Looks like 1.8 is tagged starting at 1.8.1. Will there be pre-built packages distributed Couldn't see anything new in http://files.freeswitch.org/repo/deb/freeswitch-1.8 Also - I remember discussion about Debian Stretch issues(OpenSSL related). What is the current status of Stretch and is that supposed to be the reference platform for 1.8? Thanks @FreeSWITCH team! -- Richard Chan

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