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:myu...@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. >>> >>> _______________________________________________ >>> 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 >>> >> >> >> >> -- >> 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 >> >> _______________________________________________ >> 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-tp3768542p3799274.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ > 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 _______________________________________________ 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 message @ http://n2.nabble.com/Bad-sound-quality-while-eavesdropping-tp3768542p3805109.html To unsubscribe from Re: Bad sound quality while eavesdropping, click here. -- View this message in context: http://n2.nabble.com/Bad-sound-quality-while-eavesdropping-tp3768542p3806786.html Sent from the freeswitch-users mailing list archive at Nabble.com.
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