Bad sound quality while eavesdropping
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
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>
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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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
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@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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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
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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@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
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@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@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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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> 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@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@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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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
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:
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> 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@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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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
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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@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:
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> 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@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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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@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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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:
Did you open a jira and attach all the info?
/b
On Oct 12, 2009, at 3:47 AM, Maciej Aniserowicz wrote:
Yes, I confirmed that with Wireshark (filter "rtp and ip.src == <device ip>). RTP packets are sent every 20ms.
MAniserowicz
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This bug has been now closed out in jira due to no response for requested information. If you wish to resolve this issue please follow up on your bugs when information is requested. Mike On Oct 12, 2009, at 12:04 PM, Maciej Aniserowicz wrote:
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:
Did you open a jira and attach all the info?
/b
On Oct 12, 2009, at 3:47 AM, Maciej Aniserowicz wrote:
Yes, I confirmed that with Wireshark (filter "rtp and ip.src == <device ip>). RTP packets are sent every 20ms.
MAniserowicz
participants (4)
-
Anthony Minessale
-
Brian West
-
Maciej Aniserowicz
-
Michael Jerris