[Freeswitch-users] one way speech path after attempting voice path redirecting in new invite

[Brian West]

1.4.x has been EOL for some time, you should try the latest 1.6.x release
too!


./b
On Tue, Feb 14, 2017 at 9:01 AM Bejan, Alex <alex.bejan at necam.com> wrote:

> Hello again,
>
>
>
> The following log message coincides with the initiation of the issue:
>
> 2017-02-13 12:19:52.549786 [INFO] switch_rtp.c:5867 Auto Changing port
> from 172.17.85.240:58638 to 172.17.85.240:58632
>
>
>
> Version shows: FreeSWITCH Version 1.4.26-37~64bit (-37 64bit)
>
>
>
> Tried adding in the default dialplan, under the inbound extension,  just
> before the bridge command:
>
> <action application="set" data="disable_rtp_auto_adjust=true"/>
>
> <action application="set" data="rtp_manual_rtp_bugs=accept_any_packets"/>
>
> Where is the correct place to add them?
>
> Are these parameters valid for my exact version?
>
>
>
> How can I verify if they become active? Adding a line like the one below
> shows blank.
>
> <action application="log" data="WARNING ~ Inbound disable-rtp-auto-adjust
> value:${disable-rtp-auto-adjust}"/>
>
>
>
> Thank you
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Bejan, Alex
> *Sent:* Monday, January 30, 2017 4:12 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] one way speech path after attempting
> voice path redirecting in new invite
>
>
>
> Hello,
>
>
>
> This is my first posting into this list although I monitored activities in
> the past.
>
> Freeswitch behaves great in our environment – thank you all who contribute!
>
>
>
> We started recently observing some one way speech path issues in the
> following scenario and I would like to ask for suggestions, please:
>
>
>
> We have incoming calls from provider to FS that further bridge to
> different Standard SIP  PBX.
>
> Call is established to SIP device connected on the PBX.
>
> The PBX immediately invites back attempting to redirect rtp path to start
> recording the call.
>
> FS starts sending RTP to the new destination.
>
>
>
> There are 2 different very similar PBX environments connected to the same
> FS:
>
> -          In one the PBX starts sending RTP even before the connection
> is ACKed. Recording succeeds.
>
> -          In the second one, PBX starts sending RTP almost one second
> later. It seems FS reverts back to the original speech path, while the
> other devices remain unaware, resulting in one way speech path.
>
>
>
> The rtp-timeout-sec and rtp-hold-timeout-sec are on the default values
>
>     <!-- rtp inactivity timeout -->
>
>     <param name="rtp-timeout-sec" value="300"/>
>
>     <param name="rtp-hold-timeout-sec" value="1800"/>
>
>
>
> Are there any other parameters that could have such an effect?
>
> Your suggestions on further troubleshooting or how to make this work even
> when RTP is not being received will be greatly appreciated!
>
>
>
> Thank you very much!
>
> Regards
>
>
> _________________________________________________________________________
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>
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-- 

*Brian West*
brian at freeswitch.org

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