June 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

5 7

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

3 7

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

3 2

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

2 4

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 ------------------------------------------------------------------------

4 9

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.

7 13

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

4 6

SRTP crypto selection
by Bipin Patel 21 Aug '18

21 Aug '18

hi, is it possible to force a selected crypto for SRTP as i cant seem to find a way to force a crypto to the user such that is phone uses that for media, secondly i see zoiper sends the below in SDP a=crypto:5 AES_256_CM_HMAC_SHA1_80 inline:854nJNIKSyhr0sJs+Ol+Qv2eSqC2gnydK8iVHUAXb4Xx7oW8iMM2jVi7R9xV6A== a=crypto:6 AES_256_CM_HMAC_SHA1_32 inline:854nJNIKSyhr0sJs+Ol+Qv2eSqC2gnydK8iVHUAXb4Xx7oW8iMM2jVi7R9xV6A== a=crypto:3 AES_192_CM_HMAC_SHA1_80 inline:854nJNIKSyhr0sJs+Ol+Qv2eSqC2gnydK8iVHUAXb4Xx7oW8iMM= a=crypto:4 AES_192_CM_HMAC_SHA1_32 inline:854nJNIKSyhr0sJs+Ol+Qv2eSqC2gnydK8iVHUAXb4Xx7oW8iMM= a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:854nJNIKSyhr0sJs+Ol+Qv2eSqC2gnydK8iVHUAX a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:854nJNIKSyhr0sJs+Ol+Qv2eSqC2gnydK8iVHUAX where as FS always uses the below only switch_core_media.c:1484 Found suite AES_CM_128_HMAC_SHA1_80 and also i noticed FS and zoiper have some variations in crypto names such as switch_core_media.c:1479 looking for crypto suite [AES_CM_192_HMAC_SHA1_32] in [4 AES_192_CM_HMAC_SHA1_32 and this never matches and never used at all (notice the placement of CM_192 in FS whereas zoiper has 192_CM) -- Regards, Bipin ------------------------------------------------------------------------

5 8

video on conference with PC and MAC browser works but fail with iPhone safari via webrtc
by 薛光宏 25 Jul '18

25 Jul '18

Hi All, here is our simple architecture <client A>---<FreeSWITCH> <client B>----/ - client A and B are both registered on FreeSWITCH via websocket (using sipjs 0.9.2) - we enable video-mode= transcode and load mod_av on FreeSWITCH. - if client A uses chrome on PC and client B use safari on *MAC*, they can see each other, talk to each other in the same conference. - if client A uses chrome on PC and client B use safari on *iPhone*(8plus, iOS 11.4), they can NOT see each other, talk to each other in the same conference. I though safari on MAC works, then safari on iPhone should work, but not. Is it possible to fix that with changing configuration of FreeSWITCH? or any suggestion? Thanks in advance. FreeSWITCH Version 1.6.20 Ubuntu Server Ubuntu 14.04.1 LTS 64bit libav-11.4 or libav-12 See below configurations: 1. enable video-mode= transcode <param name="conference-flags" value="transcode-video|minimize-video-encoding"/> <param name="video-mode" value="transcode"/> 2.enable VP8/H264 codec support <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMU,PCMA,VP8,H264"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,VP8,H264"/> 3.enable mod_av and disable mod_h26x <load module="mod_av"/>

2 4

Influencing "Update Callee ID" behavior
by Thilo-Alexander Ginkel 19 Jul '18

19 Jul '18

Hello everyone, when placing an external outbound call via mod_sofia I am seeing the callee's number and name being updated as soon as the called party picks up the call: 2017-12-08 10:19:24.933767 [INFO] sofia.c:1279 sofia/external/040xxx Update Callee ID to "Outbound Call" <40xxx> This seems to override what has been previously set as sip_callee_id_name (e.g., using <action application="set" data="sip_callee_id_name=${db(select/cid/${2})}"/>). There are two problems that I'd like to solve: 1. When sip_callee_id_name was previously set (e.g., via DB lookup) I'd like to keep that value (or repeat the lookup) so that the callee's name remains intact. 2. Certain destinations incorrectly strip the loeading zero from the callee's number, which I'd like to avoid displaying as is. Is there any way to influence what is displayed as name and number, i.e. keep the previously set sip_callee_id_name and apply some kind of normalization to the reported number to cope with the missing leading zero? Thanks, Thilo

1 1