That is the remote sdp, not the local sdp. They are sending ptime 20,
not us. Are they actually sending 20 ms packets or are they sending 30?
MIke
On Aug 23, 2009, at 4:20 PM, Peter P GMX wrote:
Hello Anthony,
I set p...@30i,p...@30i and I can see in the logs that PCMA is used.
However
mr fritz is lying somewhere
get a pcap of the traffic from fritz to FS and look at the size of the audio
packets
if they are 160(172 with headers) bytes then it's 20ms if it's 240 (252)
then it's 30ms
if it's saying 20 but it means 30 you should leave the last change in place
and also add in
Hello Anthony,
I set p...@30i,p...@30i and I can see in the logs that PCMA is used.
However ptime is set to 20 msec as shown in the Logs:
2009-08-23 22:11:29.530301 [DEBUG] sofia.c:3309 Remote SDP:
v=0
o=user 2075230 2075230 IN IP4 217.xx.xx.xxx
s=call
c=IN IP4 217.xx.xx.xxx
t=0 0
m=audio 7078
On Thu, 2009-08-20 at 14:35 -0700, Michael Collins wrote:
Just curious - if it seems to be working with Asterisk but not
FreeSWITCH then could you do some tcpdumps of working vs. non-working
calls and then analyze them with Wireshark? I think Jason Garland's
ClueCon presentation(s) might be
Hello Michael,
I made some tests with Freeswitch and Fritzbox and found by Wireshark that:
within one call
* Freeswitch starts sending 20msec packets, then after ~0,2 second
sends 30msec packets
* FritzBox always sends 30msec packets.
The average jitter is below 2 msec in both
Try setting that in your sip profile:
param name=rtp-autofix-timing value=false /
Thats a feature to work around with devices lying about their ptime in
their sdp payload.
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
On
Hello Mathieu,
thank for your help. But this however didn't change the behaviour.
I've read of a patch in mod_sofia.c which partly corrects the problem
temporarily:
When I change Line 784 to
if (switch_rtp_ready(tech_pvt-rtp_session) codec_ms != tech_pvt-codec_ms) {
to
if
You can ship me one whois bkw.org, I can add it to my lab.
/b
On Aug 21, 2009, at 10:38 AM, Peter P GMX wrote:
BTW: We can ship you a FritzBox if you need one for testing.
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try setting FS to 30ms too
edit vars.xml and add @30i to everywhere you see PCMU or PCMA so it looks
like p...@30i
from:
X-PRE-PROCESS cmd=set data=global_codec_prefs=g7...@32000h
,g7...@16000h,G722,PCMU,PCMA,GSM/
X-PRE-PROCESS cmd=set data=outbound_codec_prefs=PCMU,PCMA,GSM/
to:
Hello,
when calling from Fritzbox to a Snom Phone , sound is fine. But when
calling an internal Freeswitch number (conference, mailbox) i hear a
very choppy voice coming from the fritzbox side. I think it may have to
do with the ptime 20msec/30msec.
Example: When calling from the fritzbox to a
This is a bug in the fritzbox... you have to set your codec neg. to
greedy on the sofia profile and that should fix it.
/b
On Aug 20, 2009, at 8:35 AM, Peter P GMX wrote:
Hello,
when calling from Fritzbox to a Snom Phone , sound is fine. But when
calling an internal Freeswitch number
Besides taking a hammer to it? Have you tried to make sure you have
the latest firmware? Try setting the ptime on the fritz to 20ms?
I really can't trust a product that has fritz in its name... it might
go on the fritz :P pun intended.
/b
On Aug 20, 2009, at 9:27 AM, Peter P GMX wrote:
Hello Brian,
yes we have updated to the latest Fritzbox Firmware. These FritzBoxes
are widely spread here in Germany. I know of a SIP provider who has 5
Mio FritzBoxes out there. So overall I expect about 10Mio FritzBoxes in
Germany, and they are covering a big stake of in the market. So they
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