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<DIV><FONT size=2 face=Arial>Yes, I confirmed that with Wireshark (filter
"rtp and ip.src == <device ip>). RTP packets are sent every
20ms.</FONT></DIV>
<DIV><FONT size=2 face=Arial></FONT> </DIV>
<DIV><FONT size=2 face=Arial>MAniserowicz</FONT></DIV>
<DIV><FONT size=2 face=Arial></FONT> </DIV>
<BLOCKQUOTE style="BORDER-LEFT: #000000 2px solid; PADDING-LEFT: 5px; PADDING-RIGHT: 0px; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px">
<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV style="FONT: 10pt arial; BACKGROUND: #e4e4e4; font-color: black"><B>From:</B>
<a href="http://n2.nabble.com/user/SendEmail.jtp?type=node&node=3806786&i=0" target="_top" rel="nofollow">[hidden email]</a> </DIV>
<DIV style="FONT: 10pt arial"><B>To:</B> <a href="http://n2.nabble.com/user/SendEmail.jtp?type=node&node=3806786&i=1" target="_top" rel="nofollow">[hidden email]</a> </DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Monday, October 12, 2009 12:21
AM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> Re: [Freeswitch-users] Bad sound
quality while eavesdropping</DIV>
<DIV><BR></DIV>can you confirm from an rtp packet trace that they are all
really <BR>sending 20ms? <BR><BR>Mike <BR><BR>On Oct 10, 2009, at 6:04
AM, Maciej Aniserowicz wrote: <BR>
<DIV class=shrinkable-quote><div class='shrinkable-quote'><BR>> <BR>> Hi, <BR>> Here are the
messages with a:ptime parameter. All the calls are <BR>> started by
<BR>> commands sent through socket. <BR>> I'm not sure if this is all
information you need, please let me know <BR>> if <BR>> something
is missing here and I'll post that. <BR>> <BR>> 1) starting connection
with x-lite (number 2003, the eavesdropper): <BR>> <BR>> INVITE
sip:<A href="http://n2.nabble.com/user/SendEmail.jtp?type=node&node=3805109&i=0" rel="nofollow" target=_top>[hidden email]</A>:60188;rinstance=80b8f8d92af87cd2
SIP/ <BR>> 2.0 <BR>> Via: SIP/2.0/UDP <BR>>
192.168.3.159:15060;rport;branch=z9hG4bKD9meHQN5XQ88K <BR>>
Max-Forwards: 69 <BR>> From: "MyApp" <sip:<A href="http://n2.nabble.com/user/SendEmail.jtp?type=node&node=3805109&i=1" rel="nofollow" target=_top>[hidden email]</A>>;tag=jpQ6D7D2jUXvF <BR>>
To: <sip:<A href="http://n2.nabble.com/user/SendEmail.jtp?type=node&node=3805109&i=2" rel="nofollow" target=_top>[hidden
email]</A>:60188;rinstance=80b8f8d92af87cd2> <BR>> Call-ID:
663c4500-3024-122d-57a2-9be8d85a4cff <BR>> CSeq: 121465610 INVITE
<BR>> Contact: <sip:<A href="http://n2.nabble.com/user/SendEmail.jtp?type=node&node=3805109&i=3" rel="nofollow" target=_top>[hidden email]</A>:15060> <BR>>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN <BR>> Allow:
INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, <BR>> NOTIFY,
REFER, UPDATE, REGISTER, INFO, PUBLISH <BR>> Supported: timer,
precondition, path, replaces <BR>> Allow-Events: talk, presence,
dialog, call-info, sla, <BR>> include-session-description, presence.winfo,
message-summary, refer <BR>> Content-Type: application/sdp <BR>>
Content-Disposition: session <BR>> Content-Length: 447
<BR>> Remote-Party-ID: "MyApp" <BR>> <sip:<A href="http://n2.nabble.com/user/SendEmail.jtp?type=node&node=3805109&i=4" rel="nofollow" target=_top>[hidden
email]</A>>;party=calling;screen=yes;privacy=off <BR>> <BR>>
v=0 <BR>> o=FreeSWITCH 2223565947735016740 3096553520713245589 IN
IP4 <BR>> 192.168.3.159 <BR>> s=FreeSWITCH <BR>>
c=IN IP4 192.168.3.159 <BR>> t=0 0 <BR>> m=audio 29966
RTP/AVP 0 115 107 9 8 3 101 13 <BR>> a=rtpmap:0 PCMU/8000 <BR>>
a=rtpmap:115 G7221/32000 <BR>> a=fmtp:115 bitrate=48000
<BR>> a=rtpmap:107 G7221/16000 <BR>> a=fmtp:107
bitrate=32000 <BR>> a=rtpmap:9 G722/8000 <BR>> a=rtpmap:8
PCMA/8000 <BR>> a=rtpmap:3 GSM/8000 <BR>> a=rtpmap:101
telephone-event/8000 <BR>> a=fmtp:101 0-16 <BR>>
a=rtpmap:13 CN/8000 <BR>> a=ptime:20 <BR>> <BR>> <BR>> 2)
starting connection with cisco ip phone (number 2006, first leg of <BR>>
eavesdropped session): <BR>> <BR>> INVITE sip:<A href="http://n2.nabble.com/user/SendEmail.jtp?type=node&node=3805109&i=5" rel="nofollow" target=_top>[hidden email]</A>:5060;user=phone SIP/2.0 <BR>>
Via: SIP/2.0/UDP <BR>>
192.168.3.159:15060;rport;branch=z9hG4bKg40rp87FNjB1p <BR>>
Max-Forwards: 69 <BR>> From: "MyApp" <sip:<A href="http://n2.nabble.com/user/SendEmail.jtp?type=node&node=3805109&i=6" rel="nofollow" target=_top>[hidden email]</A>>;tag=Q3N2pe2K47ctS <BR>>
To: <sip:<A href="http://n2.nabble.com/user/SendEmail.jtp?type=node&node=3805109&i=7" rel="nofollow" target=_top>[hidden email]</A>:5060;user=phone> <BR>>
Call-ID: 6e6fd8a0-3024-122d-57a2-9be8d85a4cff <BR>> CSeq:
121465616 INVITE <BR>> Contact: <sip:<A href="http://n2.nabble.com/user/SendEmail.jtp?type=node&node=3805109&i=8" rel="nofollow" target=_top>[hidden email]</A>:15060> <BR>>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN <BR>> Allow:
INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, <BR>> NOTIFY,
REFER, UPDATE, REGISTER, INFO, PUBLISH <BR>> Supported: timer,
precondition, path, replaces <BR>> Allow-Events: talk, presence,
dialog, call-info, sla, <BR>> include-session-description, presence.winfo,
message-summary, refer <BR>> Content-Type: application/sdp <BR>>
Content-Disposition: session <BR>> Content-Length: 447
<BR>> Remote-Party-ID: "MyApp" <BR>> <sip:<A href="http://n2.nabble.com/user/SendEmail.jtp?type=node&node=3805109&i=9" rel="nofollow" target=_top>[hidden
email]</A>>;party=calling;screen=yes;privacy=off <BR>> <BR>>
v=0 <BR>> o=FreeSWITCH 8453497903781974949 2294419114567885490 IN
IP4 <BR>> 192.168.3.159 <BR>> s=FreeSWITCH <BR>>
c=IN IP4 192.168.3.159 <BR>> t=0 0 <BR>> m=audio 17670
RTP/AVP 0 115 107 9 8 3 101 13 <BR>> a=rtpmap:0 PCMU/8000 <BR>>
a=rtpmap:115 G7221/32000 <BR>> a=fmtp:115 bitrate=48000
<BR>> a=rtpmap:107 G7221/16000 <BR>> a=fmtp:107
bitrate=32000 <BR>> a=rtpmap:9 G722/8000 <BR>> a=rtpmap:8
PCMA/8000 <BR>> a=rtpmap:3 GSM/8000 <BR>> a=rtpmap:101
telephone-event/8000 <BR>> a=fmtp:101 0-16 <BR>>
a=rtpmap:13 CN/8000 <BR>> a=ptime:20 <BR>> <BR>> <BR>> 3)
starting connection with extension playing a file (number 9999,
<BR>> second <BR>> leg of eavesdropped session): <BR>> <BR>>
SIP/2.0 200 OK <BR>> Via: SIP/2.0/UDP <BR>>
192.168.3.159:5080;rport=5080;branch=z9hG4bKv6Dg7myj4tvBS <BR>>
From: "FreeSWITCH" <BR>>
<sip:myuser@mydomain;transport=udp>;tag=091j2Q0Fre8vp <BR>>
To: <sip:<A href="http://n2.nabble.com/user/SendEmail.jtp?type=node&node=3805109&i=10" rel="nofollow" target=_top>[hidden email]</A>:15060>;tag=U7t5Xt51rB64Q
<BR>> Call-ID: 7551b7d0-3024-122d-b3a6-156c4939c4f8 <BR>>
CSeq: 121465623 INVITE <BR>> Contact: <sip:<A href="http://n2.nabble.com/user/SendEmail.jtp?type=node&node=3805109&i=11" rel="nofollow" target=_top>[hidden email]</A>:15060;transport=udp> <BR>>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN <BR>>
Accept: application/sdp <BR>> Allow: INVITE, ACK, BYE, CANCEL,
OPTIONS, PRACK, MESSAGE, SUBSCRIBE, <BR>> NOTIFY, REFER, UPDATE, REGISTER,
INFO, PUBLISH <BR>> Supported: timer, precondition, path, replaces
<BR>> Allow-Events: talk, presence, dialog, call-info, sla, <BR>>
include-session-description, presence.winfo, message-summary, refer <BR>>
Content-Type: application/sdp <BR>> Content-Disposition:
session <BR>> Content-Length: 263 <BR>> <BR>> v=0
<BR>> o=FreeSWITCH 5614997529598779838 636912243381649698 IN IP4
<BR>> 192.168.3.159 <BR>> s=FreeSWITCH <BR>>
c=IN IP4 192.168.3.159 <BR>> t=0 0 <BR>> m=audio 30086
RTP/AVP 0 101 13 <BR>> a=rtpmap:0 PCMU/8000 <BR>>
a=rtpmap:101 telephone-event/8000 <BR>> a=fmtp:101 0-16 <BR>>
a=rtpmap:13 CN/8000 <BR>> a=ptime:20 <BR>> <BR>>
<BR>> <BR>> <BR>> Anthony Minessale wrote: <BR>>> <BR>>>
you probably have some device lying about ptime everywhere <BR>>> look
at a sip trace an pay especially close attention to ptime:x
<BR>>> param in <BR>>> sdp <BR>>> if you don't understand
this just attach it here <BR>>> <BR>>> execute the following at
the cli <BR>>> sofia profile internal siptrace on <BR>>> sofila
loglevel debug <BR>>> <BR>>> <BR>>> <BR>>> On Thu, Oct
8, 2009 at 8:27 AM, Maciej Aniserowicz < <BR>>> <A href="http://n2.nabble.com/user/SendEmail.jtp?type=node&node=3805109&i=12" rel="nofollow" target=_top>[hidden email]</A>> wrote: <BR>>>
<BR>>>> <BR>>>> It's the same on the trunk (the last rev I
used was not so old <BR>>>> anyway). <BR>>>>
<BR>>>> Codecs are the same on both legs: <BR>>>> read
codec/read rate: PCMU 8000 <BR>>>> write
codec/write rate: PCMU 8000 <BR>>>> <BR>>>> MA
<BR>>>> <BR>>>> <BR>>>> <BR>>>>
<BR>>>> Michael Jerris wrote: <BR>>>>>
<BR>>>>> What codecs are all the call legs using, also, please try
current <BR>>>>> svn <BR>>>>> trunk.
<BR>>>>> <BR>>>>> Mike <BR>>>>>
<BR>>>>> On Oct 7, 2009, at 3:39 AM, Maciej Aniserowicz wrote:
<BR>>>>> <BR>>>>>> <BR>>>>>> Sorry
about posting several questions at once, I wasn't aware it's
<BR>>>>>> "rude". <BR>>>>>> Let's concentrate on
this issue then. <BR>>>>>> <BR>>>>>> I use FS
rev 14994. Phones on extensions: <BR>>>>>> 1) x-lite
<BR>>>>>> 2) cisco sip phone <BR>>>>>> 3) audio
played by fs to the extension being eavesdropped <BR>>>>>>
<BR>>>>>> I did not change any codec configuration, I just use
the standard <BR>>>>>> one that <BR>>>>>> comes
with both FS and the phones. <BR>>>>>> Some time ago someone on
FS irc channel told me that this is just <BR>>>>>> how FS
<BR>>>>>> eavesdropping works... from your response I
understand that this <BR>>>>>> is
<BR>>>>>> not <BR>>>>>> entirely true?
<BR>>>>>> <BR>>>>>> Maciej Aniserowicz
<BR>>>>>> <BR>>>>>> <BR>>>>>>
<BR>>>>>> Anthony Minessale wrote: <BR>>>>>>>
<BR>>>>>>> That's is a somewhat vague position.
<BR>>>>>>> <BR>>>>>>> You did not mention
which version of FreeSWITCH you are <BR>>>>>>>
running, the <BR>>>>>>> phones <BR>>>>>>>
being used in your example, your configuration, the codecs in use
<BR>>>>>>> etc. <BR>>>>>>>
<BR>>>>>>> BTW, <BR>>>>>>> I think you
should only ask one question at a time on this list.
<BR>>>>>>> The list <BR>>>>>>> is run by
volunteers and it's sort of rude to expect 3 or 4
<BR>>>>>>> threads <BR>>>>>>> to be
<BR>>>>>>> tended to concerning the same one individual.
<BR>>>>>>> <BR>>>>>>>
<BR>>>>>>> 2009/10/5 Maciej Aniserowicz <<A href="http://n2.nabble.com/user/SendEmail.jtp?type=node&node=3805109&i=13" rel="nofollow" target=_top>[hidden email]</A>> <BR>>>>>>>
<BR>>>>>>>> Hello, <BR>>>>>>>> When
I use eavesdropping in FreeSWITCH, the sound quality is
<BR>>>>>>>> really bad. <BR>>>>>>>>
Is <BR>>>>>>>> there any way to improve it? Is this a
known problem? <BR>>>>>>>> Br/
<BR>>>>>>>> Maciej Aniserowicz
<BR>>>>>>>> <BR>>>>> <BR>>>>>
<BR>>>>> _______________________________________________
<BR>>>>> FreeSWITCH-users mailing list <BR>>>>> <A href="http://n2.nabble.com/user/SendEmail.jtp?type=node&node=3805109&i=14" rel="nofollow" target=_top>[hidden email]</A> <BR>>>>> <A href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" rel="nofollow" target=_top>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</A><BR>>>>>
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<BR>>>> users <BR>>>>> <A href="http://www.freeswitch.org" rel="nofollow" target=_top>http://www.freeswitch.org</A><BR>>>>>
<BR>>>>> <BR>>>> <BR>>>> -- <BR>>>>
View this message in context: <BR>>>> <A href="http://n2.nabble.com/Bad-sound-quality-while-eavesdropping-tp3768542p3788019.html" rel="nofollow" target=_top>http://n2.nabble.com/Bad-sound-quality-while-eavesdropping-tp3768542p3788019.html</A><BR>>>>
Sent from the freeswitch-users mailing list archive at Nabble.com.
<BR>>>> <BR>>>>
_______________________________________________ <BR>>>>
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<BR>>> <BR>>> <BR>>> <BR>>> -- <BR>>> Anthony
Minessale II <BR>>> <BR>>> FreeSWITCH <A href="http://www.freeswitch.org/" rel="nofollow" target=_top>http://www.freeswitch.org/</A><BR>>> ClueCon <A href="http://www.cluecon.com/" rel="nofollow" target=_top>http://www.cluecon.com/</A><BR>>> Twitter: <A href="http://twitter.com/FreeSWITCH_wire" rel="nofollow" target=_top>http://twitter.com/FreeSWITCH_wire</A><BR>>> <BR>>>
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irc.freenode.net #freeswitch <BR>>> <BR>>> FreeSWITCH Developer
Conference <BR>>> sip:<A href="http://n2.nabble.com/user/SendEmail.jtp?type=node&node=3805109&i=20" rel="nofollow" target=_top>[hidden email]</A> <sip <BR>>> %<A href="http://n2.nabble.com/user/SendEmail.jtp?type=node&node=3805109&i=21" rel="nofollow" target=_top>[hidden email]</A>> <BR>>> iax:<A href="http://n2.nabble.com/user/SendEmail.jtp?type=node&node=3805109&i=22" rel="nofollow" target=_top>[hidden email]</A>/888 <BR>>> googletalk:<A href="http://n2.nabble.com/user/SendEmail.jtp?type=node&node=3805109&i=23" rel="nofollow" target=_top>[hidden email]</A><googletalk%3Aconf <BR>>>
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<BR>>> <BR>>> _______________________________________________
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<BR>> users <BR>> <A href="http://www.freeswitch.org" rel="nofollow" target=_top>http://www.freeswitch.org</A></DIV></div><BR>_______________________________________________
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View this message in context: <a href="http://n2.nabble.com/Bad-sound-quality-while-eavesdropping-tp3768542p3806786.html">Re: [Freeswitch-users] Bad sound quality while eavesdropping</a><br>
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