[Freeswitch-users] rtp timestamp issue on pickup

[Chris Martineau]

Thanks that worked great.

 

Regards

 

Chris

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter Olsson
Sent: 21 December 2011 12:46
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] rtp timestamp issue on pickup

 

Try to enable rtp-rewrite-timestamps according to http://wiki.freeswitch.org/wiki/RTP_Issues#Dropped_Audio, that might help.

 

/Peter

 

Från: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] För Chris Martineau
Skickat: den 21 december 2011 13:33
Till: FreeSWITCH Users Help
Ämne: Re: [Freeswitch-users] rtp timestamp issue on pickup

 

Thanks for the reply,

 

Looking at the wireshark it doesn't seem to activate the mark bit again?

 

It's not that freeswitch lowers the ts it is just forwarding the incoming timestamp which is out of synch with the timestamp of the parked call, maybe this is why it is not being set?

 

Any other ideas?

 

Thanks

 

Chris

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter Olsson
Sent: 21 December 2011 09:51
To: 'freeswitch-users at lists.freeswitch.org'
Subject: Re: [Freeswitch-users] rtp timestamp issue on pickup

 

If FS lowers the timestamp FS activates the mark bit, which should cause the client to be able to handle this correctly.

 

/Peter

 

 

Från: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] För Chris Martineau
Skickat: den 21 december 2011 10:33
Till: freeswitch-users at lists.freeswitch.org
Ämne: [Freeswitch-users] rtp timestamp issue on pickup

 

Hi,

 

My specific application requires an incoming call to pickup an existing waiting call.

 

The first call is terminated and parked.

 

The second call rings in and uses intercept to then connect it to the waiting call.

 

The problem I have is that the second call has a lower rtp timestamp than the waiting call causing the client codec to wait until it catches up?

 

i.e client incoming rtp stream timestamp effectively looks like this ...

 

existing stream from freeswitch

37920

38080

38240

On intercept the timestamps then become that of the other calls stream

3040

3200

...

The client then doesn't get any incoming voice until the new timestamps catch up!

 

Is there any way to set freeswitch to maintain the timestamps?

 

Should the client allow for this?

 

Any ideas would be greatly appreciated.

 

Regards

 

Chris  

 

!DSPAM:4ef1d21e32762117429169! 

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