Re: [asterisk-users] Audio cutting in and out - asterisk 13.1 cert6 / confbridge
"timing test" does similar, it just doesn't do the automatic calculation. Confbridge normally operates at a mixing interval of 20ms, which is 50 ticks per second. That would be what you would want to test. If you don't get 50 per second then that means ConfBridge will not provide a steady source of media to each participant and it will be up to each remote jitterbuffer to handle the delayed traffic. Enough of it and stuff goes wonky. You could also see this on a packet capture. That would determine if it's timing related or not. -- Thanks Joshua. We're talking about pretty long gaps in the audio, probably around 10-15 seconds which is quite a bit of missed ticks at 20ms sampling. I was poking around the timing code trying to get a better understanding of things and found that Asterisk uses timerfd_create with CLOCK_MONOTONIC as the clock. The man page states CLOCK_MONOTONIC is affected by incremental adjustments to the time made by things like NTP. I may be completely off track here but would something like vmtools that tries to correct the clock skew (caused by VMware) be causing some issues here? Meaning that if asterisk calls timerfd_create but then the time is adjusted could that throw off the timing of the descriptor? Regards Bob This email with all information contained herein or attached hereto may contain confidential and/or privileged information intended for the addressee(s) only. If you have received this email in error, please contact the sender and immediately delete this email in its entirety and any attachments thereto. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Audio cutting in and out - asterisk 13.1 cert6 / confbridge
Hello, We use Asterisk extensively for conferencing - for the last 8 years or so this has been the 1.4/1.6/1.8 releases running chan_sip and meetme for up to around 350 concurrent users. Right around that number DAHDI hit's a hard coded memory limit and kicks allocation errors in the log. [Jun 22 10:04:13] WARNING[9095] app_meetme.c: Unable to open DAHDI pseudo channel: Cannot allocate memory In order to support our growing user count we recently upgraded to 13.1-cert6 with pjsip and replaced meetme with confbridge. During all of our UAT and load testing everything seemed to be fine, there were no perceived audio quality issues or any logs that would indicate an issue. Unfortunately now that we're in production I'm getting consistent complaints that the audio from participants is cutting in and out. It only seems to occur while under load with > 350 users but that is anecdotal at best. This is not a simple networking issue, we've pretty much ruled that out with various performance testing. That was not the case initially and we had incrementing UDP packet receive errors which we've eliminated with a bit of tuning. There are numerous architectural differences between the two installations and so far I have not been successful in determining the root cause. I'm reaching out to the community and the developers for insight and feedback hoping there is prior experience with this issue and how to resolve it. As you can see below the most significant difference is probably the use of VMware on the new install. I've tuned the ESXi host and guest per VMware's recommendations for latency and jitter (full cpu/mem reservations) with no improvement. With all of the reading I've done I suspect my issue may come down to a timing source and VMware not providing a reliable clock. It seems they allow a backlog of interrupts and if it hasn't caught up in 60s they are simply dropped. Before I rip apart the environment and rebuild on physical I'd like to try and confirm that hypothesis. In the past this was a simple matter of running dahdi_test which would report the accuracy. I'm not sure how to interpret the results of "timing test" in the Asterisk CLI. If I increase the number of ticks per second the results are erratic while under load. I'm using the timerfd module in Asterisk with a 1000HZ tick kernel and high res timers enabled. I've tried both hpet and tsc as system clock sources, both exhibit the same breaks in audio. It sounds like someone presses the mute button in the middle of a sentence. Any insight is appreciated! Here are the specs on the new install: Physical HWCisco UCS Blade (UCSB-B200-M3) vMware ESXI 5.5 VM Guest 4 vCPU w/ 32G of RAM tuned for latency/jitter (sensitivity=high) and full cpu/memory reservations. VM OS Redhat EL7 kernel 3.10.0-327.13.1.el7.x86_64 with tickless disabled e.g nohz=off and 1000HZ. Asterisk13.1-cert6 using the timerfd module. Regards Robert McGilvray SS GlobeOp Associate Director, IT Network Security GlobeOp Financial Services | 1565 Front Street | Yorktown Hts NY 10598 t: +1 (914)-293-3584 | f: +1 (914)-293-3510 rmcgi...@globeop.com | www.ssctech.com<http://www.ssctech.com/> | www.sscglobeop.com<http://www.sscglobeop.com/> Follow us: Twitter<http://twitter.com/GlobeOp> | Facebook<http://www.facebook.com/pages/SSC-Technologies-Inc/191750415876> | LinkedIn<http://www.linkedin.com/company/globeop-financial-services> This email with all information contained herein or attached hereto may contain confidential and/or privileged information intended for the addressee(s) only. If you have received this email in error, please contact the sender and immediately delete this email in its entirety and any attachments thereto. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 13.1-cert6 Now Available
> Are you selectively loading modules? If so you need the new res_pjproject.so > loaded. Yes. That did it, thanks. Bob This email with all information contained herein or attached hereto may contain confidential and/or privileged information intended for the addressee(s) only. If you have received this email in error, please contact the sender and immediately delete this email in its entirety and any attachments thereto. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 13.1-cert6 Now Available
Hello,
This build fails to load res_pjsip.so, it kicks back symbol lookup errors for
ast_pjproject_get_buildopt. Certified cert4 works fine, pjproject is 2.4.5.
[Apr 21 13:15:34] Loading res_pjsip.so.
[Apr 21 13:15:34] -- Local IPv4 address determined to be: 10.33.204.12
[Apr 21 13:15:34] -- Local IPv6 address determined to be:
[fe80::250:56ff:fe95:501a]
[Apr 21 13:15:34] == Parsing
'/home/asterisk/asterisk/certified-13.1-cert6/etc/pjsip.conf': Found
[Apr 21 13:15:34] == Manager registered action PJSIPShowEndpoints
[Apr 21 13:15:34] == Manager registered action PJSIPShowEndpoint
/home/asterisk/asterisk/certified-13.1-cert6/sbin/asterisk: symbol lookup
error: /home/asterisk/asterisk/certified-13.1-cert6/lib/modules/res_pjsip.so:
undefined symbol: ast_pjproject_get_buildopt
ykt1cfbprd1:/home/asterisk/asterisk/certified-13.1-cert6/etc# ldd
/home/asterisk/asterisk/certified-13.1-cert6/lib/modules/res_pjsip.so
linux-vdso.so.1 => (0x7ffc9aa2e000)
libpjsua2.so.2 => /usr/local/lib/libpjsua2.so.2 (0x7fe01653f000)
libstdc++.so.6 => /lib64/libstdc++.so.6 (0x7fe016235000)
libpjsua.so.2 => /usr/local/lib/libpjsua.so.2 (0x7fe015f84000)
libpjsip-ua.so.2 => /usr/local/lib/libpjsip-ua.so.2 (0x7fe015d6e000)
libpjsip-simple.so.2 => /usr/local/lib/libpjsip-simple.so.2
(0x7fe015b5b000)
libpjsip.so.2 => /usr/local/lib/libpjsip.so.2 (0x7fe015914000)
libpjmedia-codec.so.2 => /usr/local/lib/libpjmedia-codec.so.2
(0x7fe015709000)
libpjmedia-videodev.so.2 => /usr/local/lib/libpjmedia-videodev.so.2
(0x7fe015506000)
libpjmedia-audiodev.so.2 => /usr/local/lib/libpjmedia-audiodev.so.2
(0x7fe015301000)
libpjmedia.so.2 => /usr/local/lib/libpjmedia.so.2 (0x7fe0150be000)
libpjnath.so.2 => /usr/local/lib/libpjnath.so.2 (0x7fe014e9e000)
libpjlib-util.so.2 => /usr/local/lib/libpjlib-util.so.2
(0x7fe014c7b000)
libsrtp.so.2 => /usr/local/lib/libsrtp.so.2 (0x7fe014a66000)
libgsmcodec.so.2 => /usr/local/lib/libgsmcodec.so.2 (0x7fe01485a000)
libspeex.so.2 => /usr/local/lib/libspeex.so.2 (0x7fe014631000)
libilbccodec.so.2 => /usr/local/lib/libilbccodec.so.2
(0x7fe014422000)
libg7221codec.so.2 => /usr/local/lib/libg7221codec.so.2
(0x7fe01421)
libpj.so.2 => /usr/local/lib/libpj.so.2 (0x7fe013ff1000)
libm.so.6 => /lib64/libm.so.6 (0x7fe013cef000)
librt.so.1 => /lib64/librt.so.1 (0x7fe013ae6000)
libpthread.so.0 => /lib64/libpthread.so.0 (0x7fe0138ca000)
libc.so.6 => /lib64/libc.so.6 (0x7fe013509000)
libgcc_s.so.1 => /lib64/libgcc_s.so.1 (0x7fe0132f2000)
/lib64/ld-linux-x86-64.so.2 (0x7fe0169ef000)
Regards
Robert McGilvray
o: 914 293 3584
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asterisk
Development Team
Sent: Wednesday, April 20, 2016 12:04 PM
To: Asterisk Users Mailing List
Subject: [asterisk-users] Asterisk 13.1-cert6 Now Available
The Asterisk Development Team has announced the release of Certified Asterisk
13.1-cert6.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/certified-asterisk
The release of Certified Asterisk 13.1-cert6 resolves several issues reported
by the community and would have not been possible without your participation.
Thank you!
The following are the issues resolved in this release:
Bugs fixed in this release:
---
* ASTERISK-25929 - res_pjsip_registrar: AOR_CONTACT_ADDED events
not raised (Reported by Joshua Colp)
* ASTERISK-25928 - res_pjsip: URI validation done outside of PJSIP
thread (Reported by Joshua Colp)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.1-cert6
Thank you for your continued support of Asterisk!
This email with all information contained herein or attached hereto may contain
confidential and/or privileged information intended for the addressee(s) only.
If you have received this email in error, please contact the sender and
immediately delete this email in its entirety and any attachments thereto.
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[asterisk-users] Hold/Resume no audio - Asterisk 13.7.2 / PJSIP 2.4.5 / CUCM 8.6.2
Hello, We currently have a production Asterisk box running 1.8.20.1 which uses MeetMe and chan_sip for conferences. I have been testing the new versions of Asterisk with PJSIP and ConfBridge but have run into an issue which is preventing us from moving forward. Everything works fine until a call is placed on hold, after resuming the call the user cannot hear audio from the bridge. The same thing works perfectly with 1.8.20.1. The scenario is: Cisco 8845 SIP (G722) -> CUCM 8.6.2 (also tested 10.5, same issue) - > SIP trunk to Asterisk 13.7.2 PJSIP with delayed offer -> ConfBridge. Tcpdump reveals Asterisk is sending the RTP to the endpoint so I suspect we're dealing with a bug / interop issue with the culprit possibly being a=inactive lines in the SDP. I've included a link (on drive) to two separate SIP traces, one using ngrep and the other is the output of pjsip logging and the relevant sections of my pjsip.conf https://drive.google.com/folderview?id=0B6XOeEMvID0vX2FxTXNkZWlodWM=sharing Can anyone offer some insight? Regards, BobM This email with all information contained herein or attached hereto may contain confidential and/or privileged information intended for the addressee(s) only. If you have received this email in error, please contact the sender and immediately delete this email in its entirety and any attachments thereto. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hold/Resume no audio - Asterisk 13.7.2 / PJSIP 2.4.5 / CUCM 8.6.2
I did some more troubleshooting eliminating G722 just in case there was an issue with transcoding / MTP which has resulted in a slightly different SDP but resume still doesn't work. Full sip dialog: https://drive.google.com/open?id=0B6XOeEMvID0vZVdOV2NzM3NJZGM Initial call setup appears to be correct. CUCM sends an early offer to Asterisk with SDP, Asterisk responds with a 200 OK with SDP and a=sendrecv. When I place the call on HOLD the CUCM sends a delayed offer INVITE to Asterisk: Asterisk responds with a 200 OK containing the SDP with a=sendrecv CUCM ACKS with an SDP containing a=sendonly When I resume the call the CUCM sends a delayed offer INVITE to Asterisk: Asterisk responds with a 200 OK containing the SDP with a=recvonly CUCM ACKS with an SDP containing a=sendonly I may be missing or interpreting something incorrectly but that does not right for a RESUME scenario. Per RFC32645 the CUCM is responding in one of the two ways permitted: "If a stream is offered as sendonly, the corresponding stream MUST be marked as recvonly or inactive in the answer. If a media stream is listed as recvonly in the offer, the answer MUST be marked as sendonly or inactive in the answer. " Is this a bug or am I wrong in my interpretation of the dialog? Thanks! Robert McGilvray o: 914 293 3584 From: Robert McGilvray Sent: Thursday, March 17, 2016 12:55 PM To: 'asterisk-users@lists.digium.com' Subject: Hold/Resume no audio - Asterisk 13.7.2 / PJSIP 2.4.5 / CUCM 8.6.2 Hello, We currently have a production Asterisk box running 1.8.20.1 which uses MeetMe and chan_sip for conferences. I have been testing the new versions of Asterisk with PJSIP and ConfBridge but have run into an issue which is preventing us from moving forward. Everything works fine until a call is placed on hold, after resuming the call the user cannot hear audio from the bridge. The same thing works perfectly with 1.8.20.1. The scenario is: Cisco 8845 SIP (G722) -> CUCM 8.6.2 (also tested 10.5, same issue) - > SIP trunk to Asterisk 13.7.2 PJSIP with delayed offer -> ConfBridge. Tcpdump reveals Asterisk is sending the RTP to the endpoint so I suspect we're dealing with a bug / interop issue with the culprit possibly being a=inactive lines in the SDP. I've included a link (on drive) to two separate SIP traces, one using ngrep and the other is the output of pjsip logging and the relevant sections of my pjsip.conf https://drive.google.com/folderview?id=0B6XOeEMvID0vX2FxTXNkZWlodWM=sharing Can anyone offer some insight? Regards, BobM This email with all information contained herein or attached hereto may contain confidential and/or privileged information intended for the addressee(s) only. If you have received this email in error, please contact the sender and immediately delete this email in its entirety and any attachments thereto. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_confbridge production ready?
I do not use ConfBridge() in a large installation. I use MeetMe on 1.6.0.* The timing is different for ConfBridge, as it does not require DAHDI. If you have that good of an experience with 1.4, why change anything? I like new things. ConfBridge eliminates the need for an external timing source (like the Sangoma card) which allows me to run Asterisk on our preferred OS, Solaris. It also supports 16kHz audio which fits in nicely with all my Polycom wideband phones. Unfortunately I answered my own question by installing Asterisk 1.6.2.x on solaris 10 and giving it a shot. Launching ConfBridge segfaults asterisk everytime :( Thanks for the feedback. Bob -- This email with all information contained herein or attached hereto may contain confidential and/or privileged information intended for the addressee(s) only. If you have received this email in error, please contact the sender and immediately delete this email in its entirety and any attachments thereto. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] app_confbridge production ready?
I have an existing conference bridge running on Asterisk 1.4.2 using MeetMe and it's been pretty much rock solid since it was installed. We do around 460,000 minutes on it monthly and peak at about 150 simultaneous sip channels. I'm adding a second bridge for redundancy purposes into another facility and would like to go with app_confbridge and Solaris. Since it's a relatively new app I have to question if it's been put to the test and whether I can expect the same kind of stability. Does anyone use confbridge in a large installation and can provide feedback on its stability, quality in comparison to MeetMe? I use a sangoma card in my 1.4.2 box to provide timing and it has never been an issue. Can I expect similar performance from the new timing API? Thanks Bob -- This email with all information contained herein or attached hereto may contain confidential and/or privileged information intended for the addressee(s) only. If you have received this email in error, please contact the sender and immediately delete this email in its entirety and any attachments thereto. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe option question
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard
Kenner
Sent: Thursday, October 08, 2009 8:23 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] MeetMe option question
We've started to use Asterisk for conferencing and have been getting
some
complaints. Our configuration is that some people call in from home,
but
we have a physical conference room with a Polycom. When somebody was
giving
a presentation in the physical conference room, we were told that the
remote
people kept hearing him cut in and our. To me, this sounds like the
talking
optimization was getting false negatives.
Is there a way to say don't apply talk optimization to this user so we
could add that to the Polycom when it called it? From a quick scan of
app_meetme.c, I don't see one, but it doesn't look too hard to add.
--
You can do this in the dialplan. Just launch MeetMe with different
options based on the caller, I use SQL and AGI in my installations but
it doesn't have to be that complex.
If (${CALLERID(num)} = Polycom callerID) {
MeetMe(CONFROOM|AscM);
} else {
MeetMe(CONFROOM|AscMo);
}
My syntax is probably off a bit but that should get you started. You may
also want to consider just turning off the talker optimization entirely
- I've found it to be very problematic and generates more complaints
than it's worth.
Bob
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Re: [asterisk-users] CDR Records for MeetMe
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anthony
Sent: Friday, September 18, 2009 11:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] CDR Records for MeetMe
Andy Rosen wrote:
... figure out a good way to log which conference ID that is being
used.
The only way I have found to do this is in the events, the conference
enter event has the unique id of the call, which will tie it to the cdr,
and the conference number.
Hope this helps!
Anthony
---
This may not be exactly what you're looking for but it's one possible
solution. I use a wrapper around MeetMe to emulate what my users were
used to doing. I playback an intro file which prompts the user for a pin
number then use it to query a SQL db for the conference id. I then
record the entered pin in the CDR userfield. You could easily modify it
to prompt for the conference ID then store it instead. Here's an example
from my dialplan.
Read(PIN,globeop/bridge_greeting-nancy,7,,3,5);
AGI(go-meetme.agi|${PIN});
Set(CDR(UserField)=${PIN});
MeetMe(${CONFROOM}|${OPTIONS}|${PIN});
${OPTIONS} and ${CONFROOM} are populated in my AGI script.
Bob
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[asterisk-users] Fax detection on SIP channel
Is there a built-in way of detecting fax tones, or a switch to T.38 on a SIP channel? I need to periodically check some efax servers for availability and figured the best way to ensure they are operational is to check for tones. I've looked into Nvdetect but the company seems to have gone out of business and I don't want to be stuck with a solution that won't make it through an upgrade of asterisk. My ITSP supports T.38 and should send a Re-Invite so is there a way to just set a channel variable when it sees this and use that as an indicator of success? I have a sangoma card in the machine but it doesn't have any T1/E1 connections, so unless I'm mistaken I can't use the fax detection in zaptel. Bob -- This email with all information contained herein or attached hereto may contain confidential and/or privileged information intended for the addressee(s) only. If you have received this email in error, please contact the sender and immediately delete this email in its entirety and any attachments thereto. (W1) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
