- FreeSWITCH-users - FreeSWITCH Mailing Lists

[mod http cache][zero byte file]
by Bilal Abbasi 12 Jan '18

12 Jan '18

Hi Users, I am using mod http cache in freeswitch to get sound file from amazon S3, today i noticed a file downloaded with zero bytes, i tried alot to replicate the issue but could not able to get that. I was just intrested to know if someone get across this issue? Or some one using http cache in production. P.S: my system is in dev phase right now, and i am curious to know is it safe to use http cache in prod. Regards Abbasi

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Re: [Freeswitch-users] verto, private ip, stun, and Chrome
by Mundkowsky, Robert 11 Jan '18

11 Jan '18

Well, we found out commenting out the ACLs in the verto.conf.xml worked:  Seems the client's private IP was one from our LAN, so FreeSWITCH thought it was a valid request from our LAN. But I wonder if STUN should have fixed this? Robert Mundkowsky From: Mundkowsky, Robert Sent: Thursday, January 11, 2018 5:25 PM To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Subject: verto, private ip, stun, and Chrome Hi, We are using a Verto javascript that is maybe a little out of date and most recent version of Chrome. We are seeing that client's private IP address is being sent to FreeSWITCH rather than client's public IP. We have "iceServers: true" which should tell verto to use the default google STUN server. I am guessing the below code that uses "navigator.mozGetUserMedia" likely is causing the problems since getusermedia has been tweaked a lot by Chrome and Firefox (per https://developer.mozilla.org/en-US/docs/Web/API/Navigator/getUserMedia ). window.moz = !!navigator.mozGetUserMedia; function RTCPeerConnection(options) { var gathering = false, done = false; var w = window, PeerConnection = w.mozRTCPeerConnection || w.webkitRTCPeerConnection, SessionDescription = w.mozRTCSessionDescription || w.RTCSessionDescription, IceCandidate = w.mozRTCIceCandidate || w.RTCIceCandidate; var STUN = { url: !moz ? 'stun:stun.l.google.com:19302' : 'stun:23.21.150.121' }; robert ________________________________ This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. Thank you for your compliance. ________________________________

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verto, private ip, stun, and Chrome
by Mundkowsky, Robert 11 Jan '18

11 Jan '18

Hi, We are using a Verto javascript that is maybe a little out of date and most recent version of Chrome. We are seeing that client's private IP address is being sent to FreeSWITCH rather than client's public IP. We have "iceServers: true" which should tell verto to use the default google STUN server. I am guessing the below code that uses "navigator.mozGetUserMedia" likely is causing the problems since getusermedia has been tweaked a lot by Chrome and Firefox (per https://developer.mozilla.org/en-US/docs/Web/API/Navigator/getUserMedia ). window.moz = !!navigator.mozGetUserMedia; function RTCPeerConnection(options) { var gathering = false, done = false; var w = window, PeerConnection = w.mozRTCPeerConnection || w.webkitRTCPeerConnection, SessionDescription = w.mozRTCSessionDescription || w.RTCSessionDescription, IceCandidate = w.mozRTCIceCandidate || w.RTCIceCandidate; var STUN = { url: !moz ? 'stun:stun.l.google.com:19302' : 'stun:23.21.150.121' }; robert ________________________________ This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. Thank you for your compliance. ________________________________

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Codec renegotiation
by Gregor Nanger 11 Jan '18

11 Jan '18

Hi Guys! Have one question. I want to bridge call to gateway and if I get 503, bridge to another gateway. This works as expected and have no problem. But have question if I can renogotiate codecs from a leg to b when send call to another provider? Is this just matter of absolute_codec_string variable in bridge for b leg? Best regards, Gregor

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(no subject)
by Ali Haider 11 Jan '18

11 Jan '18

Hiii jeol https://freeswitch.org/confluence/pages/viewpage.action?pageId=1966780 I’m followed these steps on windows but after installation of visual studio 2013 which was may be required 182 error is found for missing files Please tell me which version of visual studio is installed to build soln and how to remove error Sent from my iPhone

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Audio distorted when Negotiation of codecs on Re-Invite (G711->G722)
by José Lopes 11 Jan '18

11 Jan '18

Hello, I have an issue with audio distorted during a call when there is a reinvite that change the audio codec from G711 to G722. The issue seems to be caused by G722 which have a diferent sample rate (16000Hz) than G711 (8000Hz). I have already reported this issue at https://freeswitch.org/jira/browse/FS-10779 . I notice that this is a complex situation, so I want to ask if there is any workarround to this situation? I need to use G722 because I need to have HD voice calls. Thanks in advance for the attention. Best regards, Jose Lopes

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Wrong jitter in the RECV_RTCP_MESSAGE event
by Alexander Haugg 11 Jan '18

11 Jan '18

Hi, Codec = PCMU Wireshark RTP analyse: Jitter in the Wireshark max=2,9 ms; mean=0,57 But the last RECV_RTCP_MESSAGE event show this: ========================================== SSRC: 0eb0e841 NTP-Most-Significant-Word: 3724642770 NTP-Least-Significant-Word: 4165677056 RTP-Timestamp: 14487886 Sender-Packet-Count: 1409 Octect-Packet-Count: 225440 Last-RTP-Timestamp: 14487924 RTP-Rate: 8000 Capture-Time: 1515653966925374 Source0-SSRC: 7c1b0ae5 Source0-Fraction: 0 Source0-Lost: 0 Source0-Loss-Avg: 0 Source0-Highest-Sequence-Number-Received: 32561 Source0-Jitter: 159 Source0-LSR: 2311973851 Source0-DLSR: 63014 And the last RTCP event from the Wireshark show this: ============================================= Real-time Transport Control Protocol (Sender Report) 10.. .... = Version: RFC 1889 Version (2) ..0. .... = Padding: False ...0 0001 = Reception report count: 1 Packet type: Sender Report (200) Length: 12 (52 bytes) Sender SSRC: 0x0eb0e841 (246474817) Timestamp, MSW: 3511979744 (0xd1548ee0) Timestamp, LSW: 2709503779 (0xa17fbf23) [MSW and LSW as NTP timestamp: Apr 16, 2011 21:55:44.630855000 UTC] RTP timestamp: 3435008682 Sender's packet count: 3257276450 Sender's octet count: 4258211364 Source 1 Identifier: 0x463e7aa2 (1178499746) SSRC contents Fraction lost: 248 / 256 Cumulative number of packets lost: 1105247 Extended highest sequence number received: 1972857173 Interarrival jitter: 1013423106 Last SR timestamp: 1007735790 (0x3c10d3ee) Delay since last SR timestamp: 108449976 (1654815 milliseconds) [RTCP frame length check: OK - 52 bytes] 1. What is the formula to get the jitter in ms from the rtcp? I can’t find anything in the web, or I can’t recognise the answer. 2. What is wrong, the RECV_RTCP_MESSAGE event or the RTCP from the receiver? If I have the answer to the first question, i can answer the second question by my self ;-) Thanks a lot!!! By the way, it is the newest FreeSwitch from yesterday.

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record_session recordings out of sync
by Mircea BH 11 Jan '18

11 Jan '18

Hello everyone, Happy new year. I have a new freeswitch production version 1.6 installed on Debian 8, 64bits. I have just two sip phones registered and one gateway (Alcatel oxo). All are on the same switch, in different vlans. I use record_session to record calls automatically. The recordings are fine for calls between the two sip phones, but the ones between sip phone and Alcatel gateway are all out of sync. I is the same regardless of settings (stereo, mono, jitterbuffer activated). Did someone had this issue and found a solution? Thank you. Best regards, Mircea.

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Odd situation bridge
by Raimundo Pérez Nieves 10 Jan '18

10 Jan '18

Hi, I have this situation: 1) Leg A calls Leg B 2) Leg B answers 3) Start exchange media but here is the problem: a) Leg B starts with silence and after that starts ringtone (but understand you are inside a bridge) The question: How can I bridge A with B only when I receive from Leg B the inside ringtone (when it answered). Consider in this situation -When Leg B answers it produces silence and after that ringtone and after that the person from the other side- . Thanks a lot

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Re: [Freeswitch-users] Problem with bridge_pre_execute_bleg_app
by Raimundo Pérez Nieves 10 Jan '18

10 Jan '18

Iam not able to solve the issue with bridge_pre_execute_bleg_app. The problem I am having is that the call is answered, is in 200 OK, and is exchanging media, but there is silence before ringtone starts. This is my trace: https://docs.google.com/document/d/1cvUybw7MTrRA032V0A2G0UiKBca_fsrPBSOR7rs… <https://docs.google.com/document/d/1cvUybw7MTrRA032V0A2G0UiKBca_fsrPBSOR7rs…> DialPlan: action application="export" data="nolocal:absolute_codec_string=G729"/> <action application="set" data="ringback=${us-ring}"/> <action application="ring_ready"/> <action application="bridge" data="{ignore_early_media=true}sofia/external/550075378355222(a)64.34.75.23"/>

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