[Freeswitch-users] Bad sound quality while eavesdropping

Maciej Aniserowicz maciej.aniserowicz at gmail.com
Sat Oct 10 03:04:04 PDT 2009


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:2003 at 192.168.3.100:60188;rinstance=80b8f8d92af87cd2 SIP/2.0
   Via: SIP/2.0/UDP 192.168.3.159:15060;rport;branch=z9hG4bKD9meHQN5XQ88K
   Max-Forwards: 69
   From: "MyApp" <sip:0000000000 at 192.168.3.159>;tag=jpQ6D7D2jUXvF
   To: <sip:2003 at 192.168.3.100:60188;rinstance=80b8f8d92af87cd2>
   Call-ID: 663c4500-3024-122d-57a2-9be8d85a4cff
   CSeq: 121465610 INVITE
   Contact: <sip:mod_sofia at 192.168.3.159:15060>
   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:0000000000 at 192.168.3.159>;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:2006 at 192.168.2.106:5060;user=phone SIP/2.0
   Via: SIP/2.0/UDP 192.168.3.159:15060;rport;branch=z9hG4bKg40rp87FNjB1p
   Max-Forwards: 69
   From: "MyApp" <sip:0000000000 at 192.168.3.159>;tag=Q3N2pe2K47ctS
   To: <sip:2006 at 192.168.2.106:5060;user=phone>
   Call-ID: 6e6fd8a0-3024-122d-57a2-9be8d85a4cff
   CSeq: 121465616 INVITE
   Contact: <sip:mod_sofia at 192.168.3.159:15060>
   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:0000000000 at 192.168.3.159>;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 at mydomain;transport=udp>;tag=091j2Q0Fre8vp
   To: <sip:9999 at 192.168.3.159:15060>;tag=U7t5Xt51rB64Q
   Call-ID: 7551b7d0-3024-122d-b3a6-156c4939c4f8
   CSeq: 121465623 INVITE
   Contact: <sip:mod_sofia at 192.168.3.159:15060;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 <
> maciej.aniserowicz at gmail.com> 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 <maciej.aniserowicz at gmail.com>
>> >>>
>> >>>> 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
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> --
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>>
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> 
> 
> 
> -- 
> Anthony Minessale II
> 
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