[Freeswitch-users] Bad sound quality while eavesdropping
Michael Jerris
mike at jerris.com
Sun Oct 11 15:14:50 PDT 2009
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:
>
> 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
>>>>>>>
>>>>
>>>>
>>>> _______________________________________________
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>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>
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>>>>
>>>>
>>>
>>> --
>>> 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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>>
>>
>>
>> --
>> Anthony Minessale II
>>
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>>
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>
> --
> View this message in context: http://n2.nabble.com/Bad-sound-quality-while-eavesdropping-tp3768542p3799274.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
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