[gst-devel] Re: patch for rtpg711pay + some questions (#325148)

Edgard Lima edgard.lima at indt.org.br
Wed Dec 28 13:13:06 CET 2005

Hi Kai, it seems to be fine pls just go ahead, commit it


Kai Vehmanen wrote:

> Hi,
> here's another bug-report plus patch:
> http://bugzilla.gnome.org/show_bug.cgi?id=325148
> The max-ptime thing is pretty clear. By using the baseclass 
> 'max-ptime' attribute, this patch makes G711/RTP usable for VoIP. 
> 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 big, even with otherwise optimal sender and receiver 
> implementations. The app could of course modify the MTU as well (via 
> basertppayload 'mtu'), but this a bit ugly... (MTU has to be specified 
> in octests, but the app doesn't know how many octets one msec of audio 
> takes for a given payloader, nor is the codec frame-based or not).
> But, but, the min-ptime is trickier. This patch adds a 10msec low 
> threshold for ptime (= how much audio is put in one packet). With 
> frameless codecs as G711, there's no lower limit (but 0) on how much 
> audio to put in packets, but obviously you don't want to be sending 
> very small packets (1ms => packets would have 8 bytes of payload, plus 
> 40 bytes of RTP/UDP/IP headers, plus link headers). RFC3551 recommends 
> a range of 0-200msec common to all audio payloads, and a default of 
> 20msec.
> Is this a sane thing to do - should other payloaders for frameless 
> codecs do the same (add a min-ptime limit)? Should this be a 
> base-class level
> attribute? For frame-based codecs, one frame is of course the natural 
> min-ptime...
> PS A tip for all devels using gst-RTP - it's best to align the
>    values of src element 'block-size' (and 'latency-time' for alsasrc),
>    and payload 'max-ptime'. This allows to minimize the delay in sending
>    RTP (.. and minimizes memcpys in gst_adapter).
> -- 
>  under work: Sofia-SIP at http://sofia-sip.sf.net

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