GNOME Bugzilla – Bug 325148
Bugs in G711 RTP packetization logic
Last modified: 2006-02-08 17:27:43 UTC
Please describe the problem:
The current code has the following problems:
- does not take baseclass max_ptime property into account
- does not have a lower limit for ptime (limit for how small packets to send
- minor bug in setting payload_len (payloadlen is given to
gst_rtp_buffer_calc_payload_len() which subtracts header size)
Note: ptime = packetization interval, i.e. how many msecs of audio to put in one
Steps to reproduce:
Does this happen every time?
Created attachment 56469 [details] [review]
Patch that fixes the described problems.
This patch makes G711/RTP usable for VoIP use. Without the patch, the payloader fills the packets with upto MTU-size of encoded data. This results in RTP/UDP packets that contain 125msec of audio each. This will in practise cause the end-to-end delay to become too large, even with otherwise optimal sender and receiver implementations.
Patch applied to CVS on 2006-01-24.