[asterisk-users] 0005162: RTP Packetization : Few questions
yusuf
yusuf at ecntelecoms.com
Fri Sep 8 02:26:14 MST 2006
Dan Austin wrote:
>>>2006-08-31 22:11:22 WARNING[1278]: frame.c:1072
>>>ast_codec_pref_getsize: Framing not set for codec alaw, using
>>>default 20
>>
>>As far as the above is concerned I have the following:
>
>
>>I am using Asterisk 1.2.10, patched with this patch for 1.2.10.
>>I have 2 * boxes. They call each other over SIP, and I have in
>>sip.conf on both boxes
>
>
>>autoframing=yes
>>disallow=all
>>allow=g729:80
>
>
>>When A calls B, it sets ptime:80.
>
>
>>On B I see this:
>>We're at 192.168.0.64 port 11004
>>Adding codec 0x100 (g729) to SDP
>>Sep 7 18:16:16 WARNING[5529]: frame.c:1072 ast_codec_pref_getsize:
>>Framing not set for codec g729, using default 20 and ptime:20
>
>
>>So B is setting packetization to 20, when it should be 80, and is
>>not respecting autoframing.
>
>
>>I have tried this with reinvites=yes and no, and autoframing=yes and
>>no, still the same.
>
>
> The autoframing patch forgot to remove an earlier check for 'ptime'
> in the SDP that would cause chan_sip to ignore the ptime value.
>
> I am working on trunk, so the line numbers may not match up, but
> near line 4748 you will should find this block of code:
>
> } else if (!strncasecmp(a, "ptime:", (size_t) 6)) {
> if (debug)
> ast_verbose("Got unsupported
> a:ptime in SDP offer \n");
> breakout = TRUE;
>
> Simply comment out the breakout = TRUE; line like this.
>
> } else if (!strncasecmp(a, "ptime:", (size_t) 6)) {
> if (debug)
> ast_verbose("Got unsupported
> a:ptime in SDP offer \n");
> /* breakout = TRUE; */
>
> That fixes up autoframing in my tests, if it works for you, I will
> prepare a proper patch.
>
Hi, I will try this. But even with autoframing=no, B still sets ptime:20.
on B in sip.conf
[sipacket]
username=sipacket
secret=sipacket
type=friend
host=dynamic
context=default
disallow=all
allow=g729:60
;autoframing=yes
;canreinvite=no
--
thanks,
yusuf
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