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[maciej.aniserowicz at ...]

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

[anthony.minessale at g...]

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@gmail.com (maciej.aniserowicz@gmail.com)>

[maciej.aniserowicz at ...]

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:

[mike at jerris.com]

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:

[maciej.aniserowicz at ...]

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:

[anthony.minessale at g...]

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@gmail.com (maciej.aniserowicz@gmail.com)> wrote:

[maciej.aniserowicz at ...]

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@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@192.168.3.159>;tag=jpQ6D7D2jUXvF
To: <sip:2003@192.168.3.100:60188;rinstance=80b8f8d92af87cd2>
Call-ID: 663c4500-3024-122d-57a2-9be8d85a4cff
CSeq: 121465610 INVITE
Contact: <sip:mod_sofia@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@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@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@192.168.3.159>;tag=Q3N2pe2K47ctS
To: <sip:2006@192.168.2.106:5060;user=phone>
Call-ID: 6e6fd8a0-3024-122d-57a2-9be8d85a4cff
CSeq: 121465616 INVITE
Contact: <sip:mod_sofia@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@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@mydomain;transport=udp>;tag=091j2Q0Fre8vp
To: <sip:9999@192.168.3.159:15060>;tag=U7t5Xt51rB64Q
Call-ID: 7551b7d0-3024-122d-b3a6-156c4939c4f8
CSeq: 121465623 INVITE
Contact: <sip:mod_sofia@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:

[mike at jerris.com]

can you confirm from an rtp packet trace that they are all really
sending 20ms?

Mike

On Oct 10, 2009, at 6:04 AM, Maciej Aniserowicz wrote:

[maciej.aniserowicz at ...]

[brian at freeswitch.org]

Did you open a jira and attach all the info?

/b

On Oct 12, 2009, at 3:47 AM, Maciej Aniserowicz wrote:

[maciej.aniserowicz at ...]

Nope, I wanted to make sure that this is indeed a bug. I opened an issue in
JIRA before regarding some other matter and it turned out to be my mistake,
so I decided to try mailing list first this time.
MA



Brian West wrote: