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@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Bejan, Alex
Sent: Monday, January 30, 2017 4:12 PM
To: freeswitch-users@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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