[asterisk-users] RTP timestamp modification during SIP video call

[Dan Julius]

Hi,

I'm using asterisk 1.14.19

I'm making a video call between two SIP end-points, using h263p and iLBC. I
notice the video is jumpy and I believe the cause is due to RTP timestamps.

The sending device is working at 8fps and correctly increases the timestamp
by 11250 every frame. It appears that Asterisk is modifying the timestamps
and generating new ones which sometimes increase by 11250, but sometimes
have much longer delays.

Should asterisk use the senders RTP timestamp?
Maybe someone can explain why this is happening?


Thanks,
Dan
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