After an evaluation, GNOME has moved from Bugzilla to GitLab. Learn more about GitLab.
No new issues can be reported in GNOME Bugzilla anymore.
To report an issue in a GNOME project, go to GNOME GitLab.
Do not go to GNOME Gitlab for: Bluefish, Doxygen, GnuCash, GStreamer, java-gnome, LDTP, NetworkManager, Tomboy.
Bug 325148 - Bugs in G711 RTP packetization logic
Bugs in G711 RTP packetization logic
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:
Reported: 2005-12-28 15:08 UTC by Kai Vehmanen
Modified: 2006-02-08 17:27 UTC
See Also:
GNOME target: ---
GNOME version: ---

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
- 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:

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.