[Freeswitch-users] Bad sound quality while eavesdropping

Maciej Aniserowicz maciej.aniserowicz at gmail.com
Mon Oct 12 01:47:36 PDT 2009


Yes, I confirmed that with Wireshark (filter "rtp and ip.src == <device ip>). RTP packets are sent every 20ms.

MAniserowicz

  ----- Original Message ----- 
  From: Michael Jerris (via Nabble) 
  To: Maciej Aniserowicz 
  Sent: Monday, October 12, 2009 12:21 AM
  Subject: Re: [Freeswitch-users] Bad sound quality while eavesdropping


  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:[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 at 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 
  >>>>>>> 
  >>>> 
  >>>> 
  >>>> _______________________________________________ 
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  >>>> 
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  >>>> 
  >>>> 
  >>> 
  >>> -- 
  >>> View this message in context: 
  >>> http://n2.nabble.com/Bad-sound-quality-while-eavesdropping-tp3768542p3788019.html
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  >> 
  >> 
  >> 
  >> -- 
  >> Anthony Minessale II 
  >> 
  >> FreeSWITCH http://www.freeswitch.org/
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  >> 
  >> AIM: anthm 
  >> MSN:[hidden email] <MSN 
  >> %[hidden email]> 
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  >> %[hidden email]> 
  >> IRC: irc.freenode.net #freeswitch 
  >> 
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  >> %[hidden email]> 
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  >> googletalk:[hidden email]<googletalk%3Aconf 
  >> %[hidden email]> 
  >> pstn:213-799-1400 
  >> 
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  > 
  > -- 
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  > 
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