[Freeswitch-users] FreeSWITCH 1.8.2 - 1-2 second dropped Audio\RTP at the start of a call

Thu Dec 20 15:21:33 UTC 2018

Hi Shaun,

Can you share with the community what you found out?

I’m curious what can be the cause for that extra 2-3s delay in RTP

On Thu, Dec 20, 2018 at 01:52 Shaun Stokes <shaun.stokes at itec-support.co.uk>
wrote:

> We've identified the cause and have now implemented a fix, this was a
> local issue caused by a bug in our build script which applies tweaks to
> mod_sofia.
>
>
> Thanks,
>
> Shaun
> ------------------------------
> *From:* FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org>
> on behalf of Shaun Stokes <shaun.stokes at itec-support.co.uk>
> *Sent:* 19 December 2018 14:07:18
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] FreeSWITCH 1.8.2 - 1-2 second dropped
> Audio\RTP at the start of a call
>
>
> Thank you all for your recommendations. We raised a JIRA but it was closed
> as incomplete, we will look at re-opening once we've compiled the relevant
> logs and pcaps etc unless we reach a conclusion before hand.
>
>
> Some additional points:
>
> - We do not set "answer_delay".
>
> - There is still a 1-2 second delay after
> setting "disable_rtp_auto_adjust" on the SIP profile.
>
>
> We've identified this as a package issue, when FreeSWITCH is built with
> certain packages installed this is causing mod_sofia.so to be much smaller
> than it should be (5.49MB instead of 8.58MB).
>
>
> We're able to consistently reproduce the following when running
> FreeSWITCH on the same Debian 9 server using the same FreeSWITCH configuration
> and scripts etc:
>
> - FreeSWITCH 1.8.2 built on Debian 9 - 1-2 second audio\RTP delay on
> answer
>
> - FreeSWITCH 1.6.20 built on Debian 9 - 1-2 second audio\RTP delay on
> answer
>
> - FreeSWITCH 1.6.20 built on Debian 8 (moved binaries to Debian 9 server)
> - No audio\RTP delay on answer (audio\RTP is instantaneous)
>
> - FreeSWITCH 1.8.2 built on Debian 9 base (no additional packages) - No
> audio\RTP delay on answer (audio\RTP is instantaneous)
>
>
> Will post back once we've identified the root cause.
>
>
> Thanks,
>
> Shaun
> ------------------------------
> *From:* FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org>
> on behalf of Anthony Minessale <anthony.minessale at gmail.com>
> *Sent:* 17 December 2018 20:55
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] FreeSWITCH 1.8.2 - 1-2 second dropped
> Audio\RTP at the start of a call
>
> Sounds like a broken record but the mailing list is not the best place to
> report issues.  Almost inevitably, questions will be asked and data will
> need to be collected like logs etc.
> This is why we already point to JIRA to file tickets.
>
> Recommendations:
>
> 1) Make sure you don't have "answer_delay" set.
> 2) Check the media signaling data to make sure you are not using
> "rtp-auto-adjust which adds a time to media establishment to correct for
> incorrect media IPs.
> 3) Get a pcap as well as debug + sip trace before reporting any issue
> because its always going to be the first request anyway.
> 4) Use JIRA, feel free to ask about the JIRA here but don't rely on 1990's
> listserv server to track the issue progress.
>
> sofia global siptrace on
> console loglevel debug
> fsctl debug_level 10
>
>
>
>
> On Mon, Dec 17, 2018 at 2:48 PM Shaun Stokes <
> shaun.stokes at itec-support.co.uk> wrote:
>
> We reverted back to FreeSWITCH 1.6.20 but when this is compiled on
> the Debian 9 server the problem still occurs.
>
>
> We had to workaround some build errors for FS 1.6.20 to compile on Debian
> 9 with PostgreSQL 11 but the problem was still present, as follows.
>
> --------------------
>
> # Uninstall 1.1.0 SSL header files (libssl-dev) and install the older ones
> (libssl1.0-dev).
> apt-get install libssl1.0-dev
>
> # Fix PGSQL 11 Support
> In the file:
> /usr/src/freeswitch/srcswitch_pgsql.c
> On line 389, replace this:
> #if POSTGRESQL_MAJOR_VERSION >= 9 && POSTGRESQL_MINOR_VERSION >= 2
> With:
> #if (POSTGRESQL_MAJOR_VERSION == 9 && POSTGRESQL_MINOR_VERSION >= 2) ||
> POSTGRESQL_MAJOR_VERSION > 9
>
> # Do not build mod_flite or mod_enum
> sed -i /usr/src/freeswitch/modules.conf -e
> s:'asr_tts/mod_flite:#asr_tts/mod_flite:'
> sed -i /usr/src/freeswitch/modules.conf -e
> s:'applications/mod_enum:#applications/mod_enum:'
> --------------------
>
> We took the FS 1.6.20 binaries (pre-compiled) from a Debian 8 server and
> restored them to our Debian 9 server which resolved the issue but we had to
> copy some missing libs from a Debian 8 server:
> /usr/lib/x86_64-linux-gnu/libssl.so.1.0.0
> /usr/lib/x86_64-linux-gnu/libcrypto.so.1.0.0
>
> Given that the problem changes when the source-code is compiled on
> different servers we suspect this may be a package problem not specific to
> FreeSWITCH.
>
> This is also a problem on master, raised JIRA:
> https://freeswitch.org/jira/browse/FS-11572
>
> ------------------------------
> *From:* FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org>
> on behalf of Shaun Stokes <shaun.stokes at itec-support.co.uk>
> *Sent:* 14 December 2018 13:16:14
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] FreeSWITCH 1.8.2 - 1-2 second dropped
> Audio\RTP at the start of a call
>
> Correction, we had moved FreeSWITCH 1.4 (not 1.8) to Server 1 which worked
> without any audio delays. Upon testing FreeSWITCH 1.8 on Server 1 there is
> a 1-2 second delay before RTP is established once the call is answered.
>
> This is a FreeSWITCH 1.8.2 issue, not a Debian 9 specific (also occurs on
> Debian 8). FreeSWITCH 1.6 and 1.4 are not effected using the same
> configuration through-out.
> ------------------------------
> *From:* Shaun Stokes
> *Sent:* 14 December 2018 11:44:18
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] FreeSWITCH 1.8.2 - 1-2 second dropped
> Audio\RTP at the start of a call
>
> We have built two test servers side by side on the same hardware with the
> same configuration, as follows.
> Server 1: Debian 8 with FreeSWITCH 1.6.20
> Server 2: Debian 9 with FreeSWITCH 1.8.2
>
> We can replicate the 1-2 second delay on Server 2 only, whereas Server 1
> provides near instant RTP in both directions upon answer. Interestingly, if
> we move FreeSWITCH 1.8.2 from Server 2 to Server 1 there are still no
> issues with delay on Server 1, the problem is only observable on the Server
> 2 running Debian 9 so the problem is not specifically related to FreeSWITCH
> 1.8.2.
>
> At this stage it seems likely the issue lies with Debian 9 or the change
> in packages on Debian 9.
>
> Thanks,
> Shaun
> ------------------------------
> *From:* FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org>
> on behalf of Shaun Stokes <shaun.stokes at itec-support.co.uk>
> *Sent:* 11 December 2018 15:28:33
> *To:* FreeSWITCH Users Help
> *Subject:* [Freeswitch-users] FreeSWITCH 1.8.2 - 1-2 second dropped
> Audio\RTP at the start of a call
>
>
> Hi All,
>
>
> Since we've been using FreeSWITCH 1.8.2 we've noticed that the first 1-2
> seconds of Audio\RTP at the start of the call when the call is answered is
> now dropped\missing but this doesn't occur on 1.6.20. When comparing the
> examples we've noticed the call flow is slightly different, as follows.
>
>
> FreeSWITCH 1.8.2
>
> Leg A: switch_channel.c:2249 (sofia/internal/SRC_EXT at DOMAIN:PORT)
> Callstate Change DOWN -> RINGING
> Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering
> state [calling][0]
> Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering
> state [proceeding][180]
> Leg B: sofia.c:7401 Ring-Ready sofia/internal/DST_EXT at LAN_IP:PORT!
> Leg B: switch_channel.c:3354 (sofia/internal/DST_EXT at LAN_IP:PORT)
> Callstate Change DOWN -> RINGING
> Leg A: switch_ivr_originate.c:1246 Sending early media
> Leg A: switch_channel.c:3482 (sofia/internal/SRC_EXT at DOMAIN:PORT)
> Callstate Change RINGING -> EARLY
> Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering
> state [early][183]
> Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering
> state [completing][200]
> Leg B: switch_channel.c:3482 (sofia/internal/DST_EXT at LAN_IP:PORT)
> Callstate Change RINGING -> EARLY
> Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering
> state [ready][200]
> Leg B: sofia.c:8429 Channel [sofia/internal/DST_EXT at LAN_IP:PORT] has been
> answered
> Leg B: switch_channel.c:3781 (sofia/internal/DST_EXT at LAN_IP:PORT)
> Callstate Change EARLY -> ACTIVE
> Leg A: switch_ivr_bridge.c:766 Channel [sofia/internal/SRC_EXT at DOMAIN:PORT]
> has been answered
> Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering
> state [completed][200]
> Leg A: switch_channel.c:3781 (sofia/internal/SRC_EXT at DOMAIN:PORT)
> Callstate Change EARLY -> ACTIVE
> Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering
> state [ready][200]
> Leg A: switch_ivr_async.c:1500 No silence detection configured; assuming
> start of speech
> Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering
> state [calling][0]
> Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering
> state [calling][0]
> Leg A: sofia.c:7291 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering
> state [ready][200]
> Leg A: sofia.c:8272 Processing updated SDP
> Leg B: sofia.c:7291 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering
> state [ready][200]
>
>
> FreeSWITCH 1.6.20
>
> Leg A: switch_channel.c:2249 (sofia/internal/SRC_EXT at DOMAIN:PORT)
> Callstate Change DOWN -> RINGING
> Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering
> state [calling][0]
> Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering
> state [proceeding][180]
> Leg B: sofia.c:7192 Ring-Ready sofia/internal/DST_EXT at LAN_IP:PORT!
> Leg B: switch_channel.c:3346 (sofia/internal/DST_EXT at LAN_IP:PORT)
> Callstate Change DOWN -> RINGING
> Leg A: switch_ivr_originate.c:1215 Sending early media
> Leg A: switch_channel.c:3474 (sofia/internal/SRC_EXT at DOMAIN:PORT)
> Callstate Change RINGING -> EARLY
> Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering
> state [early][183]
> Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering
> state [completing][200]
> Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering
> state [ready][200]
> Leg A: switch_channel.c:3773 (sofia/internal/SRC_EXT at DOMAIN:PORT)
> Callstate Change EARLY -> ACTIVE
> Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering
> state [completed][200]
> Leg A: switch_ivr_originate.c:3705 Originate Resulted in Success:
> [sofia/internal/DST_EXT at LAN_IP:PORT]
> Leg B: switch_channel.c:3773 (sofia/internal/DST_EXT at LAN_IP:PORT)
> Callstate Change RINGING -> ACTIVE
> Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering
> state [ready][200]
> Leg B: Channel sofia/internal/DST_EXT at LAN_IP:PORT entering state
> [ready][200]
> Leg A: switch_ivr_async.c:1500 No silence detection configured; assuming
> start of speech
> Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering
> state [calling][0]
> Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering
> state [calling][0]
> Leg A: sofia.c:7084 Channel sofia/internal/SRC_EXT at DOMAIN:PORT entering
> state [ready][200]
> Leg A: sofia.c:8061 Processing updated SDP
> Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering
> state [completing][200]
> Leg B: sofia.c:7084 Channel sofia/internal/DST_EXT at LAN_IP:PORT entering
> state [ready][200]
>
>
> On 1.6.20 Leg B changes straight from RINGING to ACTIVE, but on 1.8.2 Leg
> B first changes from RINGING to EARLY then EARLY to ACTIVE, not sure if
> this could be related.
>
>
> We've experimented with the following to no avail.
> rtp-rewrite-timestamps
> send_silence_when_idle
> fsctl sync_clock
> suppress_cng
> ignore_early_media
>
> As per:
> https://freeswitch.org/confluence/display/FREESWITCH/RTP+Issues
> https://freeswitch.org/confluence/display/FREESWITCH/VAD+and+CNG
> https://freeswitch.org/confluence/display/FREESWITCH/send_silence_when_idle
> https://freeswitch.org/confluence/display/FREESWITCH/Early+Media
>
>
> The calls are local between two extensions\endpoints on the same
> FreeSWITCH instance and the same SIP profile, the SIP profiles on both
> servers (1.6.20 and 1.8.2) are identical.
>
>
> Does anyone have any ideas?
>
>
> Thanks,
>
> Shaun
> _________________________________________________________________________
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>
>
>
> --
> Anthony Minessale II
> Founder, FreeSWITCH.
> http://freeswitch.com
>
>
> https://youtu.be/l_hOxzCt6X4
> https://www.youtube.com/watch?v=oAxXgyx5jUw
> https://www.youtube.com/watch?v=9XXgW34t40s
> https://www.youtube.com/watch?v=NLaDpGQuZDA
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
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