February 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

FreeSWITCH Friday FreeForAll Reminder!
by Ken Rice 13 Feb '15

13 Feb '15

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888@conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon

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RTP delay with cng
by Jose Fco. Irles Durá 13 Feb '15

13 Feb '15

Hi, I have a FreeSWITCH 1.4.15 (from debian repositories), but I've been having a problem for a long time (with older versions too) and I don't know if it is a configuration problem or a bug. The FreeSWITCH runs on an openvz machine and this server is not overloaded. Before testing with trunk branch I want to share my problem looking for help to solve it (it's in production) I have an issue with some providers that send to my FreeSWITCH CN packets. When this happens, the other call leg has a terrible delay and bad quality sound.It starts with a little delay, but during the call the delay increases more and more. Capturing with wireshark and decoding rtp stream I can see 31% of packets dropped by Jitter Buff in the leg in wich the delay occurs (from FreeSWITCH IP to the other endpoint) I don't have any problem when the provider doesn't send CN packets. If I configure FreeSWITCH not to negotiate "comfort noise" (profile: supress-cng=true), the provider sends cn packets too. I tried with no luck variables such as: - send_silence_when_idle (with values "400" and "true") - bridge_generate_comfort_noise=true The rtp related params of my sofia profile are: rtp-timer-name=soft inbound-zrtp-passthru=false (don't use zrtp) rtp-timeout-sec=300 rtp-hold-timeout-sec=1800 inbound-late-negotiation=true NDLB-allow-bad-iananame=true supress-cng=true vad=none t38-passthru=true inbound-codec-prefs=PCMA,PCMU,G722,GSM outbound-codec-prefs=PCMA,PCMU,G722,GSM In the dialplan the only variables that I set related with rtp are: set inherit_codec=true export absolute_codec_string=${ep_codec_string} Thanks in advance -- Jose Fco. Irles Durá

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Responding to INVITE with 180 vs 183
by Ben Hood 12 Feb '15

12 Feb '15

Hi all, I'm wondering what influences Freeswitch's signalling response when it responds to an INVITE. In some instances it appears to follow this sequence: Peer -> INVITE (SDP) -> FS Peer <- 100 <- FS Peer <- 183 (SDP) <- FS Peer <- 200 (SDP) <- FS Peer -> ACK -> FS Which leads to a successful bridge between two legs. But in other instances it appears to follow the sequence: Peer -> INVITE (SDP) -> FS Peer <- 100 <- FS Peer <- 180 (SDP) <- FS Peer <- 200 (SDP) <- FS (1st attempt) Peer <- 200 (SDP) <- FS (2nd attempt) .... Peer <- 200 (SDP) <- FS (nth attempt) ....timeout I'm not 100% sure whether the remote peer (itself a SIP trunk) doesn't like the 100/180/200 sequence (as opposed to 100/183/200), so I need to find that out for myself. But I am wondering what factors influence Freeswitch's decision of what sequence signals to send in response to an INVITE. Any pointers appreciated, Ben

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Strange intermittent issue with SIP signals over WebRTC
by Oleg Stolyar 12 Feb '15

12 Feb '15

Hi guys, Sorry for a long email. I ran into this strange problem a couple of time over the last year where my Freeswitch stops receiving or sending SIP signals from and to WebRTC (JsSip) profile. At the same time, SIP signals are coming through just fine on a different profile which is regular SIP. It lasted for about 15 min. When the problem was over, I saw a bunch of BYE signals being sent to the WebRTC users at about the same for all the calls that ended during these 15 min. Also, INVITEs from these users that were sent during the 15 minutes actually got through (or at least showed in the logs) after the problem cleared. I have a fairly old FS - master from July 2014 but was wondering if anyone else had this problem. There are no errors in the logs and I cannot reproduce this at will. One possibility is that the network connection is somehow holding up these signals rather than FS. So, I was wondering when during the process of sending SIP signals over WebRTC does it get recorded in the log? Is it when the sending is successful or as soon as it's attempted? This could tell me whether FS doesn't even try to send the signal during this problem period or whether it's trying but cannot get through to the network somehow. Another piece of information - calls that were connected before the problem started continued just fine and the media kept coming through.

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ISDN G.722 dropouts - jitterbuffer underruns
by Thomas Auge 12 Feb '15

12 Feb '15

We're bridging webRTC to ISDN with FreeTDM and are experiencing frequent audio dropouts. On the webRTC end we've tried Opus and G.722 even with a client right next to the server. The effects are the same, so I'm fairly certain the problem is either on the ISDN end or inside FreeTDM. Unfortunately the only thing I could find that's related to the issue is a jitterbuffer underrun when this happens. I've checked every troubleshooting hint Sangoma provides and played around with every config option I could find that might even be remotely related. But other than the buffer underruns I got nothing, though that feels more like a consequence rather than the reason. There are no frame errors/losses on the wanpipe devices (ifconfig) either. Does anyone have an idea what else to check? (Freeswitch version is current master) Jitterbuffer debug: 655: I: DEBUG len:5/10 i=533/1000 - g:537 c:533 b:0 - ts:2274276068/14214225/2274275908 - m:0(0.000000%) j:1.90% js:20/0 dr:-160/0 920: O: DEBUG 2274276068:14214225 0 907: DEBUG MISSING 2274276228:14214226 1533 2274276068 160 5 1 806: DEBUG OUT QUEUE: ------------ 809: DEBUG 2274275108:14214219 r:1 809: DEBUG 2274275268:14214220 r:1 809: DEBUG 2274275428:14214221 r:1 809: DEBUG 2274275588:14214222 r:1 809: DEBUG 2274275748:14214223 r:1 811: DEBUG 812: DEBUG IN QUEUE: ------------ 815: DEBUG 2274275908:14214224 r:1 815: DEBUG 2274276068:14214225 r:1 817: DEBUG 746: DEBUG OUT QUEUE EMPTY, SWAPPING 980: DEBUG PLC 1/10 1 160 2274276228:14214226 600: DEBUG TOO LATE !!! 2274276228 833: DEBUG JITTERBUFFER NOT READY: IGNORING FRAME 655: I: DEBUG len:5/10 i=534/1000 - g:0 c:533 b:1 - ts:2274276388/14214227/2274276228 - m:1(0.100000%) j:1.90% js:20/0 dr:-160/0 655: I: DEBUG len:5/10 i=535/1000 - g:0 c:534 b:1 - ts:2274276548/14214228/2274276228 - m:1(0.100000%) j:1.90% js:20/0 dr:-160/0 833: DEBUG JITTERBUFFER NOT READY: IGNORING FRAME 655: I: DEBUG len:5/10 i=536/1000 - g:0 c:535 b:1 - ts:2274276708/14214229/2274276228 - m:1(0.100000%) j:1.90% js:20/0 dr:-160/0 833: DEBUG JITTERBUFFER NOT READY: IGNORING FRAME 655: I: DEBUG len:5/10 i=537/1000 - g:0 c:536 b:1 - ts:2274276868/14214230/2274276228 - m:1(0.100000%) j:1.90% js:20/0 dr:-160/0 833: DEBUG JITTERBUFFER NOT READY: IGNORING FRAME 655: I: DEBUG len:5/10 i=538/1000 - g:0 c:537 b:1 - ts:2274277028/14214231/2274276228 - m:1(0.100000%) j:1.90% js:20/0 dr:-160/0 920: O: DEBUG 2274276388:14214227 0

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Forwarded Voicemail transcription
by Sean Devoy 12 Feb '15

12 Feb '15

Hi all, A customer has asked if there is a way to forward an email transcription of a voicemail when it is FORWARDED from another user? They like the transcripts I deliver when voicemail arrives, and would like the same feature on forwarded voicemails. Is there somewhere this could be implemented? Thanks, Sean

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Mod_oreka and orkaudio problem.
by Volkan Oransoy 12 Feb '15

12 Feb '15

Hi all, I am trying to setup mod_oreka with orkaudio 1.7 but I couldn't. Freeswitch with mod_oreka sends packet via UDP but I couldn't manage to make orkaudio to listen to UDP ports. Is there a point that I miss? I couldn't find anything on the web. Thanks in advance, /Volkan Here is my confix.xml for orkaudio. <config> <AudioOutputPath>/var/log/orkaudio/audio</AudioOutputPath> <CapturePlugin>libvoip.so</CapturePlugin> <CapturePluginPath>/usr/lib</CapturePluginPath> <StorageAudioFormat>gsm</StorageAudioFormat> <DeleteNativeFile>yes</DeleteNativeFile> <TrackerHostname>localhost</TrackerHostname> <TrackerTcpPort>8080</TrackerTcpPort> <CapturePortFilters>LiveMonitoring</CapturePortFilters> <TapeProcessors>BatchProcessing, Reporting</TapeProcessors> <BatchProcessingEnhancePriority>true</BatchProcessingEnhancePriority> <AudioFileOwner>tomcat</AudioFileOwner> <AudioFileGroup>tomcat</AudioFileGroup> <AudioFilePermissions>644</AudioFilePermissions> <VoIpPlugin> <PcapSocketBufferSize>8388608</PcapSocketBufferSize> </VoIpPlugin> </config>

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mod_event_socket userauth command
by Борисов, Дмитрий / Dmitriy Borisov 12 Feb '15

12 Feb '15

Hi, All! I found very interesting thing while inspecting sources of mod_event_multicast.c: userauth command. As I understood, it allows to use API without system password if we add esl-password param to user definition in directory. Params esl-allowed-log, esl-allowed-events and esl-allowed-api responding for appropriate parameters of freeswitch behavior for this connection. My question is about next: can user affect on other domains in multitenant configuration if it connected to event_socket with userauth command? Can I restrict it actions to only his domain? -- with best regards, Dmitriy Borisov

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Error xml_curl
by Ing. Antonyam 12 Feb '15

12 Feb '15

> Hi, > > I have an issue in xml_curl module. If the user tries calling to PSTN, the call is not routed and the code generates an error similar when the calling plan is not set for PSTN extensions. I've set the plan in the database with "/d" but it doesn't work. If I use a static way inside the xml files works fine. > > My calling plan is working when it calls among extensions. > > I've validated that sofia status is setting with calling plan for PSTN. > > actions = sofia/gateway/${effective_caller_id_numer}/$1 > > > Do you have any idea why it's not working when I configure it in the database? > > Regards.

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Jitterbuffer debug output
by Thomas Auge 11 Feb '15

11 Feb '15

Could someone be so kind as to translate these abbreviations for me or point me to some documentation? I couldn't find anything about it ... 655: I: DEBUG len:3/7 i=843/1000 - g:789 c:842 b:0 - ts:1674565284/10466033/1674564804 - m:2(0.200000%) j:4.60% js:20/0 dr:0/-240 920: O: DEBUG 1674564964:10466031 0 Also: What exactly does the max drift config entail? How much difference in packet timing it will handle? If so, is there a good reason to keep this value significantly below the total buffer size like in the examples? Thanks! Thomas

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