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Bug 325148 - Bugs in G711 RTP packetization logic
Bugs in G711 RTP packetization logic
Status: RESOLVED FIXED
Product: GStreamer
Classification: Platform
Component: gst-plugins-good
git master
Other All
: Normal normal
: 0.10.2
Assigned To: GStreamer Maintainers
GStreamer Maintainers
Depends on:
Blocks:
 
 
Reported: 2005-12-28 15:08 UTC by Kai Vehmanen
Modified: 2006-02-08 17:27 UTC
See Also:
GNOME target: ---
GNOME version: ---

Attachments
Patch that fixes the described problems. (2.53 KB, patch)
2005-12-28 15:08 UTC, Kai Vehmanen 		none Details | Review

Description Kai Vehmanen 2005-12-28 15:08:04 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
forward)
- 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
packet


Steps to reproduce:
1. 
2. 
3. 


Actual results:


Expected results:


Does this happen every time?


Other information:
Comment 1 Kai Vehmanen 2005-12-28 15:08:42 UTC 
Created attachment 56469 [details] [review]
Patch that fixes the described problems.
Comment 2 Kai Vehmanen 2005-12-28 15:59:01 UTC 
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.
Comment 3 Kai Vehmanen 2006-01-24 18:08:38 UTC 
Patch applied to CVS on 2006-01-24.