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ADD 2 ICE CANDIDATES mod_verto
by Jazmin Marina Florez Marino 24 Apr '20

24 Apr '20

Hello all, I am working with mod_verto in a scenario where I need to send public and private IP addresses as ICE candidates. When I modify the ext-rtp-ip parameter, it only sends one option as typ host ice candidate. However, for my implementation I need to send both. What I need is the combination of those ice candidates, something like this: a=candidate:8228103366 1 udp 659136 190.84.118.88 31028 typ host generation 0 a=candidate:3496438569 1 udp 659136 192.168.1.170 24210 typ host generation 0 Is it possible to add both ice candidates in the same SDP from configuration files? Thanks Jazmin Florez Marino +57 304 564 0193 Jazmin.marino(a)on24.com SDP with local IP 2020-04-24 14:13:49.395900 [DEBUG] mod_verto.c:2502 Local SDP verto.rtc/5715976: v=0 o=FreeSWITCH 1587731419 1587731420 IN IP4 192.168.1.170 s=FreeSWITCH c=IN IP4 192.168.1.170 t=0 0 a=msid-semantic: WMS t9mERteVPrwT3aZbexPlp2U6Cspg78dq m=audio 24210 UDP/TLS/RTP/SAVPF 111 110 a=rtpmap:111 opus/48000/2 a=fmtp:111 useinbandfec=1; minptime=10; stereo=1; sprop-stereo=1 a=rtpmap:110 telephone-event/48000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=fingerprint:sha-256 FD:DF:48:02:C3:F8:D9:57:F8:C8:97:CD:3A:AB:6A:A6:97:4A:BA:A5:48:5A:3A:2D:0F:18:14:EC:D5:A9:21:86 a=setup:active a=rtcp-mux a=rtcp:24210 IN IP4 192.168.1.170 a=ice-ufrag:mScCKXydfOjcMntK a=ice-pwd:Depc4PrcDqdqfgirZlTz8iCe a=candidate:3496438569 1 udp 659136 192.168.1.170 24210 typ host generation 0 a=end-of-candidates a=ssrc:3132020181 cname:7pVPJ0AphX8bXYd1 a=ssrc:3132020181 msid:t9mERteVPrwT3aZbexPlp2U6Cspg78dq a0 a=ssrc:3132020181 mslabel:t9mERteVPrwT3aZbexPlp2U6Cspg78dq a=ssrc:3132020181 label:t9mERteVPrwT3aZbexPlp2U6Cspg78dqa0 m=video 0 UDP/TLS/RTP/SAVPF 19 2020-04-24 14:13:49.395900 [NOTICE] mod_dptools.c:1360 Channel [verto.rtc/5715976] has been answered SDP with Public IP <param name="ext-rtp-ip" value="190.84.118.88"/> 2020-04-24 14:17:23.715620 [DEBUG] mod_verto.c:2502 Local SDP verto.rtc/5715976: v=0 o=FreeSWITCH 1587724815 1587724816 IN IP4 190.84.118.88 s=FreeSWITCH c=IN IP4 190.84.118.88 t=0 0 a=msid-semantic: WMS 8gumHFWDC0sD12iPvVhSeEhUfLdXQBVO m=audio 31028 UDP/TLS/RTP/SAVPF 111 110 a=rtpmap:111 opus/48000/2 a=fmtp:111 useinbandfec=1; minptime=10; stereo=1; sprop-stereo=1 a=rtpmap:110 telephone-event/48000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=fingerprint:sha-256 FD:DF:48:02:C3:F8:D9:57:F8:C8:97:CD:3A:AB:6A:A6:97:4A:BA:A5:48:5A:3A:2D:0F:18:14:EC:D5:A9:21:86 a=setup:active a=rtcp-mux a=rtcp:31028 IN IP4 190.84.118.88 a=ice-ufrag:BiYFM6pSItFoUq2t a=ice-pwd:wTz7x9b00ImLonRPvGl8CBRb a=candidate:8228103366 1 udp 659136 190.84.118.88 31028 typ host generation 0 a=end-of-candidates a=ssrc:2930644107 cname:If1gLNJdD1HvUPbg a=ssrc:2930644107 msid:8gumHFWDC0sD12iPvVhSeEhUfLdXQBVO a0 a=ssrc:2930644107 mslabel:8gumHFWDC0sD12iPvVhSeEhUfLdXQBVO a=ssrc:2930644107 label:8gumHFWDC0sD12iPvVhSeEhUfLdXQBVOa0 m=video 0 UDP/TLS/RTP/SAVPF 19

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Wrong number of tries with originate_retries?
by René Weiss 24 Apr '20

24 Apr '20

Hi, The documentation for originate_retries says "Number of retries before giving up on originating a call (default is 0)." Because of this I would expect that: - originate_retries=0 = 1 call - originate_retries=1 = 2 calls (1 try and 1 retry) - originate_retries=2 = 3 calls (1 try and 2 retries) -originate_retries=3 = 4 calls (1 try and 3 retries) But in my testing I get: - originate_retries=0 = 1 call - originate_retries=1 = 1 call - originate_retries=2 = 2 calls - originate_retries=3 = 3 calls My originate command is: originate{originate_timeout=5,originate_retries=<X>,originate_retry_sleep_ms=5000}user/662 &park() And my FreeSWITCH version is: FreeSWITCH Version 1.10.2-release-14-f7bdd3845a~64bit (-release-14-f7bdd3845a 64bit) Am I doing something wrong here, do I understand the documentation wrong, or is the problem in freeswitch / the freeswitch documentation? Thanks.

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Auth rejected after upgrade to 1.10
by David P 23 Apr '20

23 Apr '20

A few questions I've asked before... 1) We're evaluating an upgrade to 1.10 from 1.8, and we found that verto can wss login but calls trigger "code": -32000 authentication required. What can we do to fix this? 2) We sometimes get MEDIA_TIMEOUT from our automated peer in a 1-on-1 conference. Can we use execute_on_media_timeout to attempt a new conference with the peer? 3) Can FS be made to provide TCP ice candidates in its answer sdp? (That is, does it support ICE TCP?) For users behind firewalls that block many UDP ports, this would allow avoiding added cost of TURN TCP. Cheers, David

1 0

Add body to INFO reply
by David Villasmil 23 Apr '20

23 Apr '20

Hello all, Is it possible to add an xml body (or whatever) to an incoming INFO request? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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Device / extension state (status) functionality at Freeswitch or get User (Peer) / Gateway (Trunk) state
by CpServiceSPb . 22 Apr '20

22 Apr '20

There is Freeswitch 1.8.5 built from sources worked on x64 Linux. I need to check / get internal users / trunck gateways state BEFORE using it to perform different actions in response of got conditions: 1. the called number is not exists in the directory at all; 2. the called number exist in the directory, but not registered at the time, for example don' t launch softphone; 3. the called number is inuse; 4. the called number is busy; 5. the called number is not answered due timeout 6. the called number is answered. Is there functionality looks like device and / or extension (user) state at Freeswitch as for example <dialstatus> or <channelstarus> ? How is the best way to get these different states for internal users, for each one and /or for trunk gateways, per each also ?

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send_dtmf only sends last 2 digits
by Pete Kay 22 Apr '20

22 Apr '20

Hi I used the following command to send DTMF to send to a device: send_dtmf WWWWWW98717288077WW597177 The device only can pick up "77". Could someone please tell me why and how to resolve it ? Thanks, Pete

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Join us for SIP Chronicles Live #1, featuring Giovanni Maruzzelli
by Maxim Sobolev 21 Apr '20

21 Apr '20

Dear Real-Time Friends and Colleagues! As many of you we have been totally devastated that we will have no chance to see you in the next few months to come. :-/ Some people in the community believe it might be years. I don’t necessarily agree with that opinion myself. Over the course of the last few years our team had a great time extending live coverage for some of those events that have been affected, got some experience and equipment. Instead of just waiting for the virus to clear, we decided to organize a series of bi-weekly live casts with some of the speakers that we have hoped to see at those events presenting their latest developments live and then answering questions from the audience. So without further ado, let me introduce our first guest Giovanni Maruzzelli, who is going to introduce his newest project SaraPhone ( https://github.com/gmaruzz/saraphone) Join us this Saturday, April 25th 4:30pm UTC and get a chance to ask Giovanni a question about his project live: https://youtu.be/mF9elIcVGE8 Or if you miss that opportunity, you can always watch the recording later on Sippy Labs channel on YouTube and email Giovanni your question at < gmaruzz(a)gmail.com>. SaraPhone is a bare bone SIP WebRTC phone, complete with most features real companies want to use in real world: HotDesking, Redial, BLFs, MWI, DND, PhoneBook, Hold, Transfer, Mute, Attended Transfer, Notifications, running on all Browsers both on Desktop and SmartPhone. SaraPhone is fully integrated with FusionPBX, the full-featured domain based multi-tenant PBX and voice switch for FreeSwitch. Based on SIP.js, SaraPhone works with all WebRTC compliant SIP proxies, gateways, and servers (Asterisk, OpenSIPS, Kamailio, Janus, etc). Initial author is Giovanni Maruzzelli, and SaraPhone gets its name from Giovanni's wife, Sara Hosseini. In addition to providing all of the usual DeskPhone functionality, SaraPhone got: - Desktop Notification for Incoming Calls - Live MWI update - Real Time BLFs status update - BLF click to call - Caller Name and Number Display - Call Error Cause Display - AutoAnswer - Network Disconnect Reload - Show and Set Caller-ID (incoming-outbound) Stay healthy, optimistic and productive! Also share, like and subscribe. See you soon!!! Regards, Max

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Mod avmd
by Vishal Pai 20 Apr '20

20 Apr '20

Hello All Hope you are well and safe. I am being playing around mod_avmd. I am originating calls by originate command both are External calls on DID's through gateway. Is it possible to detect the avmd on both the legs A and B using that above module. Currently only one leg A is being detected. Thank You Vishal P.

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Freeswitch drops calls after 32 seconds
by MikeKulls 20 Apr '20

20 Apr '20

I've installed freeswitch on a Raspberry PI 3 with pretty much bog stock config. The only change I've done is to delete the ipv6 profiles and changed the default password. I've connected a few different clients, 3 soft phones and 2 ATAs. When making calls with some of the clients the calls are dropped after 32 seconds. For other clients the calls will proceed for half an hour no issues. I have absolutely no complexity here, it's all internal phones on the local /24 lan. There are no firewalls, traffic only goes via a switch. I have no external SIP gateway, none of the clients are behind a NAT. I've done extensive googling and everything I find talks about NAT issues and VPNs, multiple subnets etc. I'm not running any of that. Packet capture and more details below. Some of the things I've tried - complete OS and freeswitch reinstall on the PI - disable IPv6 on the PI (made a big improvement, now the remote phone rings instantly instead of long delay) - full OS update before installing freeswitch - tried clients with both UDP and TCP - install freeswitch on CentOS 7. - Disable SIP ALG on router (traffic not going through router anyway) - Tried all devices on a different network switch - Tried client options like "use rport" etc What works: - Cisco SPA3102 ATA - Voiper latest version on Windows What doesn't work - Latest Voiper on android - Older version of voiper on windows - Billion 7404VGP - Android version of Voip by antisip TShark capture of a call dropping. IPs are: 192.168.1.245 Freeswitch 192.168.1.225 Windows PC with older copy of voiper 192.168.1.202 Cisco ATA 1 0.000000000 192.168.1.225 â 192.168.1.245 SIP/SDP 1038 Request: INVITE sip:1004@192.168.1.245;transport=UDP | 2 0.005573031 192.168.1.245 â 192.168.1.225 SIP 917 Status: 407 Proxy Authentication Required | 3 0.011167520 192.168.1.225 â 192.168.1.245 SIP 408 Request: ACK sip:1004@192.168.1.245;transport=UDP | 4 0.016217586 192.168.1.225 â 192.168.1.245 SIP/SDP 1314 Request: INVITE sip:1004@192.168.1.245;transport=UDP | 5 0.037157271 192.168.1.245 â 192.168.1.225 SIP 413 Status: 100 Trying | 6 0.094830064 192.168.1.245 â 192.168.1.202 SIP/SDP 1283 Request: INVITE sip:1004@192.168.1.202:5083;transport=tcp | 7 0.115792718 192.168.1.202 â 192.168.1.245 SIP 365 Status: 100 Trying | 8 0.115932040 192.168.1.245 â 192.168.1.202 TCP 54 5060 â 5083 [ACK] Seq=1230 Ack=312 Win=63784 Len=0 9 0.124784941 192.168.1.202 â 192.168.1.245 SIP 526 Status: 180 Ringing | 10 0.124863118 192.168.1.245 â 192.168.1.202 TCP 54 5060 â 5083 [ACK] Seq=1230 Ack=784 Win=63784 Len=0 11 0.151217241 192.168.1.245 â 192.168.1.225 SIP/SDP 1205 Status: 183 Session Progress | 12 2.263254019 192.168.1.202 â 192.168.1.245 SIP/SDP 901 Status: 200 OK | 13 2.263406934 192.168.1.245 â 192.168.1.202 TCP 54 5060 â 5083 [ACK] Seq=1230 Ack=1631 Win=63784 Len=0 14 2.272446033 192.168.1.245 â 192.168.1.202 SIP 471 Request: ACK sip:1004@192.168.1.202:5083;transport=tcp | 15 2.284211777 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 200 OK | 16 2.469876726 192.168.1.202 â 192.168.1.245 TCP 60 5083 â 5060 [ACK] Seq=1631 Ack=1647 Win=16000 Len=0 17 2.784653094 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 200 OK | 18 3.785916037 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 200 OK | 19 5.787085997 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 200 OK | 20 8.903231371 192.168.1.245 â 192.168.1.225 TCP 54 5060 â 56460 [ACK] Seq=1 Ack=1 Win=501 Len=0 21 8.903781836 192.168.1.225 â 192.168.1.245 TCP 60 [TCP ACKed unseen segment] 56460 â 5060 [ACK] Seq=1 Ack=2 Win=8212 Len=0 22 9.788128841 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 200 OK | 23 13.788284085 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 200 OK | 24 17.788442453 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 200 OK | 25 20.361180947 192.168.1.225 â 192.168.1.245 UDP 60 5060 â 5060 Len=4 26 21.788537538 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 200 OK | 27 25.788690954 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 200 OK | 28 29.789847278 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 200 OK | 29 32.583226442 192.168.1.245 â 192.168.1.202 TCP 54 [TCP Keep-Alive] 5060 â 5083 [ACK] Seq=1646 Ack=1631 Win=63784 Len=0 30 32.584076227 192.168.1.202 â 192.168.1.245 TCP 60 [TCP Keep-Alive ACK] 5083 â 5060 [ACK] Seq=1631 Ack=1647 Win=16000 Len=0 31 33.790019961 192.168.1.245 â 192.168.1.225 SIP/SDP 1169 Status: 200 OK | 32 34.284637050 192.168.1.245 â 192.168.1.225 SIP 704 Request: BYE sip:1002@192.168.1.225:5060;transport=UDP | 33 34.327728390 192.168.1.245 â 192.168.1.202 SIP 672 Request: BYE sip:1004@192.168.1.202:5083;transport=tcp | 34 34.344437951 192.168.1.202 â 192.168.1.245 SIP 381 Status: 200 OK | 35 34.393228415 192.168.1.245 â 192.168.1.202 TCP 54 5060 â 5083 [ACK] Seq=2265 Ack=1958 Win=63784 Len=0 36 34.422931003 192.168.1.225 â 192.168.1.245 SIP 466 Status: 200 OK | 37 38.352414375 192.168.1.225 â 192.168.1.245 TCP 60 [TCP ACKed unseen segment] 56460 â 5060 [PSH, ACK] Seq=1 Ack=2 Win=8212 Len=4 38 38.352610259 192.168.1.245 â 192.168.1.225 TCP 56 [TCP Previous segment not captured] 5060 â 56460 [PSH, ACK] Seq=2 Ack=5 Win=501 Len=2 39 38.393526512 192.168.1.225 â 192.168.1.245 TCP 60 [TCP ACKed unseen segment] 56460 â 5060 [ACK] Seq=5 Ack=4 Win=8212 Len=0 -- This email has been checked for viruses by AVG. https://www.avg.com

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verto and conference for webrtc audio mcu
by Roberto Della Pasqua 16 Apr '20

16 Apr '20

Hello, after a lot researching about the best viable webrtc audio mcu conference server, I have discovered freeswitch, and seems excellent. Can somebody explain me how to: 1. Setup a mcu audio conference with webrtc clients, dynamically opening and closing rooms (no need of signaling, this will be managed outside) 2. How to compile opus codec for maximum speed, eg. FFIXED instead of float (negligible audio difference, but large scaling) 3. Lib_verto and lib_conference are needed, allright? 4. I like manage the freeswitch with lib_socket, does you have examples perhaps? Thank you for this amazing software and for help with hints for best config webrtc audio only mcu. Kind regards. Roberto Della Pasqua www.dellapasqua.com<http://www.dellapasqua.com>

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