>
> Hi,
> Here are the
messages with a:ptime parameter. All the calls are
> started by
> commands sent through socket.
> I'm not sure if this is all
information you need, please let me know
> if
> something
is missing here and I'll post that.
>
> 1) starting connection
with x-lite (number 2003, the eavesdropper):
>
> INVITE
sip:
[hidden email]:60188;rinstance=80b8f8d92af87cd2
SIP/
> 2.0
> Via: SIP/2.0/UDP
>
192.168.3.159:15060;rport;branch=z9hG4bKD9meHQN5XQ88K
>
Max-Forwards: 69
> From: "MyApp" <sip:
[hidden email]>;tag=jpQ6D7D2jUXvF
>
To: <sip:
[hidden
email]:60188;rinstance=80b8f8d92af87cd2>
> Call-ID:
663c4500-3024-122d-57a2-9be8d85a4cff
> CSeq: 121465610 INVITE
> Contact: <sip:
[hidden email]:15060>
>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN
> Allow:
INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY,
REFER, UPDATE, REGISTER, INFO, PUBLISH
> Supported: timer,
precondition, path, replaces
> Allow-Events: talk, presence,
dialog, call-info, sla,
> include-session-description, presence.winfo,
message-summary, refer
> Content-Type: application/sdp
>
Content-Disposition: session
> Content-Length: 447
> Remote-Party-ID: "MyApp"
> <sip:
[hidden
email]>;party=calling;screen=yes;privacy=off
>
>
v=0
> o=FreeSWITCH 2223565947735016740 3096553520713245589 IN
IP4
> 192.168.3.159
> s=FreeSWITCH
>
c=IN IP4 192.168.3.159
> t=0 0
> m=audio 29966
RTP/AVP 0 115 107 9 8 3 101 13
> a=rtpmap:0 PCMU/8000
>
a=rtpmap:115 G7221/32000
> a=fmtp:115 bitrate=48000
> a=rtpmap:107 G7221/16000
> a=fmtp:107
bitrate=32000
> a=rtpmap:9 G722/8000
> a=rtpmap:8
PCMA/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:101
telephone-event/8000
> a=fmtp:101 0-16
>
a=rtpmap:13 CN/8000
> a=ptime:20
>
>
> 2)
starting connection with cisco ip phone (number 2006, first leg of
>
eavesdropped session):
>
> INVITE sip:
[hidden email]:5060;user=phone SIP/2.0
>
Via: SIP/2.0/UDP
>
192.168.3.159:15060;rport;branch=z9hG4bKg40rp87FNjB1p
>
Max-Forwards: 69
> From: "MyApp" <sip:
[hidden email]>;tag=Q3N2pe2K47ctS
>
To: <sip:
[hidden email]:5060;user=phone>
>
Call-ID: 6e6fd8a0-3024-122d-57a2-9be8d85a4cff
> CSeq:
121465616 INVITE
> Contact: <sip:
[hidden email]:15060>
>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN
> Allow:
INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY,
REFER, UPDATE, REGISTER, INFO, PUBLISH
> Supported: timer,
precondition, path, replaces
> Allow-Events: talk, presence,
dialog, call-info, sla,
> include-session-description, presence.winfo,
message-summary, refer
> Content-Type: application/sdp
>
Content-Disposition: session
> Content-Length: 447
> Remote-Party-ID: "MyApp"
> <sip:
[hidden
email]>;party=calling;screen=yes;privacy=off
>
>
v=0
> o=FreeSWITCH 8453497903781974949 2294419114567885490 IN
IP4
> 192.168.3.159
> s=FreeSWITCH
>
c=IN IP4 192.168.3.159
> t=0 0
> m=audio 17670
RTP/AVP 0 115 107 9 8 3 101 13
> a=rtpmap:0 PCMU/8000
>
a=rtpmap:115 G7221/32000
> a=fmtp:115 bitrate=48000
> a=rtpmap:107 G7221/16000
> a=fmtp:107
bitrate=32000
> a=rtpmap:9 G722/8000
> a=rtpmap:8
PCMA/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:101
telephone-event/8000
> a=fmtp:101 0-16
>
a=rtpmap:13 CN/8000
> a=ptime:20
>
>
> 3)
starting connection with extension playing a file (number 9999,
> second
> leg of eavesdropped session):
>
>
SIP/2.0 200 OK
> Via: SIP/2.0/UDP
>
192.168.3.159:5080;rport=5080;branch=z9hG4bKv6Dg7myj4tvBS
>
From: "FreeSWITCH"
>
<sip:myuser@mydomain;transport=udp>;tag=091j2Q0Fre8vp
>
To: <sip:
[hidden email]:15060>;tag=U7t5Xt51rB64Q
> Call-ID: 7551b7d0-3024-122d-b3a6-156c4939c4f8
>
CSeq: 121465623 INVITE
> Contact: <sip:
[hidden email]:15060;transport=udp>
>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN
>
Accept: application/sdp
> Allow: INVITE, ACK, BYE, CANCEL,
OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY, REFER, UPDATE, REGISTER,
INFO, PUBLISH
> Supported: timer, precondition, path, replaces
> Allow-Events: talk, presence, dialog, call-info, sla,
>
include-session-description, presence.winfo, message-summary, refer
>
Content-Type: application/sdp
> Content-Disposition:
session
> Content-Length: 263
>
> v=0
> o=FreeSWITCH 5614997529598779838 636912243381649698 IN IP4
> 192.168.3.159
> s=FreeSWITCH
>
c=IN IP4 192.168.3.159
> t=0 0
> m=audio 30086
RTP/AVP 0 101 13
> a=rtpmap:0 PCMU/8000
>
a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
>
a=rtpmap:13 CN/8000
> a=ptime:20
>
>
>
>
> Anthony Minessale wrote:
>>
>>
you probably have some device lying about ptime everywhere
>> look
at a sip trace an pay especially close attention to ptime:x
>> param in
>> sdp
>> if you don't understand
this just attach it here
>>
>> execute the following at
the cli
>> sofia profile internal siptrace on
>> sofila
loglevel debug
>>
>>
>>
>> On Thu, Oct
8, 2009 at 8:27 AM, Maciej Aniserowicz <
>>
[hidden email]> wrote:
>>
>>>
>>> It's the same on the trunk (the last rev I
used was not so old
>>> anyway).
>>>
>>> Codecs are the same on both legs:
>>> read
codec/read rate: PCMU 8000
>>> write
codec/write rate: PCMU 8000
>>>
>>> MA
>>>
>>>
>>>
>>>
>>> Michael Jerris wrote:
>>>>
>>>> What codecs are all the call legs using, also, please try
current
>>>> svn
>>>> trunk.
>>>>
>>>> Mike
>>>>
>>>> On Oct 7, 2009, at 3:39 AM, Maciej Aniserowicz wrote:
>>>>
>>>>>
>>>>> Sorry
about posting several questions at once, I wasn't aware it's
>>>>> "rude".
>>>>> Let's concentrate on
this issue then.
>>>>>
>>>>> I use FS
rev 14994. Phones on extensions:
>>>>> 1) x-lite
>>>>> 2) cisco sip phone
>>>>> 3) audio
played by fs to the extension being eavesdropped
>>>>>
>>>>> I did not change any codec configuration, I just use
the standard
>>>>> one that
>>>>> comes
with both FS and the phones.
>>>>> Some time ago someone on
FS irc channel told me that this is just
>>>>> how FS
>>>>> eavesdropping works... from your response I
understand that this
>>>>> is
>>>>> not
>>>>> entirely true?
>>>>>
>>>>> Maciej Aniserowicz
>>>>>
>>>>>
>>>>>
>>>>> Anthony Minessale wrote:
>>>>>>
>>>>>> That's is a somewhat vague position.
>>>>>>
>>>>>> You did not mention
which version of FreeSWITCH you are
>>>>>>
running, the
>>>>>> phones
>>>>>>
being used in your example, your configuration, the codecs in use
>>>>>> etc.
>>>>>>
>>>>>> BTW,
>>>>>> I think you
should only ask one question at a time on this list.
>>>>>> The list
>>>>>> is run by
volunteers and it's sort of rude to expect 3 or 4
>>>>>> threads
>>>>>> to be
>>>>>> tended to concerning the same one individual.
>>>>>>
>>>>>>
>>>>>> 2009/10/5 Maciej Aniserowicz <
[hidden email]>
>>>>>>
>>>>>>> Hello,
>>>>>>> When
I use eavesdropping in FreeSWITCH, the sound quality is
>>>>>>> really bad.
>>>>>>>
Is
>>>>>>> there any way to improve it? Is this a
known problem?
>>>>>>> Br/
>>>>>>> Maciej Aniserowicz
>>>>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>>
[hidden email] >>>>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>>>>
>>> UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-
>>> users
>>>>
http://www.freeswitch.org>>>>
>>>>
>>>
>>> --
>>>
View this message in context:
>>>
http://n2.nabble.com/Bad-sound-quality-while-eavesdropping-tp3768542p3788019.html>>>
Sent from the freeswitch-users mailing list archive at Nabble.com.
>>>
>>>
_______________________________________________
>>>
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>>>
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>>> users
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>>
>>
>>
>> --
>> Anthony
Minessale II
>>
>> FreeSWITCH
http://www.freeswitch.org/>> ClueCon
http://www.cluecon.com/>> Twitter:
http://twitter.com/FreeSWITCH_wire>>
>>
AIM: anthm
>> MSN:
[hidden email] <MSN
>> %
[hidden email]>
>>
GTALK/JABBER/PAYPAL:
[hidden email]<PAYPAL
>> %
[hidden email]>
>> IRC:
irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer
Conference
>> sip:
[hidden email] <sip
>> %
[hidden email]>
>> iax:
[hidden email]/888
>> googletalk:
[hidden email]<googletalk%3Aconf
>>
%
[hidden email]>
>> pstn:213-799-1400
>>
>> _______________________________________________
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>>
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>>
>
> --
> View this message in context:
http://n2.nabble.com/Bad-sound-quality-while-eavesdropping-tp3768542p3799274.html>
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>
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