[Freeswitch-users] set the Mark bit in the live audio RTP stream

[Ken Rice]

Hey Moishe,

This is a perfect one for Jira ( https://jira.freeswitch.org )  so that the
devs can track and fix it

K


On 3/27/14 6:02 PM, "Moishe Grunstein" <max at nysolutions.com> wrote:

> Hi All,
> How can tell Freeswitch to set the mark bit when going from early media to
> media? I am having a one way audio issue for the 1st 20 seconds with calls
> originating from a Cyberdata door box. FS Version 1.2.21
>  
> Below is my response from Cyberdata Support.
>  
> Hello Moishe,
>  
> I really appreciate your time working with me today attempting to get the
> intercom registered!  I know I said I would just touch bases with you
> tomorrow, but I think I have resolved the issue on my end after talking to our
> engineers and also figured out what is happening with the intercoms
> experiencing delayed audio at the intercom¹s speaker during calls.
>  
> Recently, we resolved a similar issue with a FreeSWITCH provider who reported
> the same symptoms with their intercoms.  Their FreeSWITCH server also
> transitions from early media (where the ringback tone plays to the calling
> device) to live audio just like your FreeSWITCH implementation.
>  
> We learned that our device wasn¹t properly identifying the Mark bit in the RTP
> stream and synchronizing the audio stream.  A fix was implemented and the
> audio delay was resolved with v9.1.1b01.
>  
> In your FreeSWITCH implementation, we are seeing the same symptoms but in your
> case FreeSWITCH does not set the Mark bit in the live audio RTP stream.  It
> does not change the SSRC either.  Instead, the timestamp jumps which causes
> the intercom to start recording missed timestamps until it hits a threshold
> after about 20 seconds, and after this point it resynchronizes the audio
> stream and the intercom starts playing it out of the speaker.
>  
> In the attached trace you provided to us, frame 198 shows the transition from
> early media to live audio.  The marker bit is not set.   I also included a
> .jpg highlighting frame 184 and 198 for your reference.  You will see the
> timestamp jumps but the SSRC/sequence numbers don¹t change and the mark bit is
> not set to indicate a new stream.
>  
> Based on this information, we are recommending you consult with your
> FreeSWITCH engineers and advise setting the Mark bit in the RTP stream when
> transitioning from early media to live audio since the SSRC and sequence
> numbers of the audio stream do not change.    Any calls with the Mark bit set
> in the new audio stream when switching from early media to live audio should
> play immediately without delay.  However, for calls with new audio streams
> without a Mark bit set, audio will be delayed until this is resolved in
> FreeSWITCH. 
>  
> I hope this information is helpful, Moishe!  Could you please keep us posted?
>  
> We would be happy to get on a conference call with you if it will help.  We
> know of another FreeSWITCH reseller in your area who had a similar issue so
> you are not alone. J
>  
>  
> Many thanks,
> Christina
>  
>  
>  
> Thanks,
>  
> Moishe Grunstein
> Tornado Computer Systems, Inc.
> 212.400.7650 888.IPPBX.US
> Service Request Email: support at nysolutions.com
> <mailto:support at nysolutions.com>
> Polycom Certified VAR
> Microsoft Small Business Specialist, Cisco SMB Select Certified
>  <http://www.nysolutions.com/>
> Computer Networking * Managed Services * IP Video Surveillance * Network
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>  
> 
> 
> _________________________________________________________________________
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> http://www.freeswitchsolutions.com
> 
> 
> 
> 
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-- 
Ken
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