Hi, we have a strange problem with white-noise with calls going through the ACD. We tested various setups and SipXecs versions (stable to source repository), always with the same result.
The setup: VoIP Gateway (sip.qsc.de) <----> Internet <-- --> Patton SN4638 <-- --> SipXecs (with ACD) <-- --> Polycom Soundstation IP 560 phones / Snom 300 phone (only for testing) I configured the phones to use various codecs, in the beginning the G729/G722, later PCMU,PCMA. If no hotline is called, the sound is clear, no white-noise, absolutely brilliant. As soon as I call a hotline I hear a white-noise while ringing. When I pick up the phone, the white-noise disappears after ~ 3 seconds. If I transfer the call from the Polycom Soundstation to another phone, I hear the white-noise again for the whole call. With the Snom I haven't that problem. I did a SIP trace, changed to DEBUG log level. I assume that I may have a codec problem. When calling the hotline, the ACD always answers with the following settings (sipxacd.log): sipxacd:"sipxConfigSetAudioCodecPreferences number of Codec = 3 for hInst=0x91ae058" sipxacd:"sipxConfigSetAudioCodecPreferences: PCMU PCMA audio/telephone-event" sipxacd:"sipXmediaFactoryImpl::buildCodecFactory: sReferences = PCMU PCMA audio/telephone-event with NumReject 0" sipxacd:"sipXmediaFactoryImpl::buildCodecFactory: supported codecs = PCMU PCMA audio/telephone-event with NumReject 0" sipxacd:"sipxConfigSetVideoCodecPreferences hInst=0x91ae058 bandWidth=3" sipxacd:"sipXmediaFactoryImpl::buildCodecFactory: sReferences = PCMU PCMA audio/telephone-event with NumReject 0" sipxacd:"sipXmediaFactoryImpl::buildCodecFactory: supported codecs = PCMU PCMA audio/telephone-event with NumReject 0" sipxacd:"sipxConfigSetVideoCodecPreferences:" sipxacd:"sipxConfigSetVideoCodecPreferences: Setting 3 failed, falling back to preference 0" sipxacd:"sipxConfigSetVideoCodecPreferences hInst=0x91ae058 bandWidth=0" sipxacd:"sipXmediaFactoryImpl::buildCodecFactory: sReferences = PCMU PCMA audio/telephone-event with NumReject 0" sipxacd:"sipXmediaFactoryImpl::buildCodecFactory: supported codecs = PCMU PCMA audio/telephone-event with NumReject 0" I changed the settings for codecs in the mediaserver-config, sipxpark-config, freeswitch.xml + sipX_profile.xml. But still I get always the log entries from above. Here'a SIP trace from the SNOM 300 phone calling a hotline: ________________________________ Sent to udp:192.168.150.19:5060 at 17/4/2009 02:46:41:836 (1074 bytes): INVITE sip:[email protected];user=phone SIP/2.0 v: SIP/2.0/UDP 192.168.150.158:2052;branch=z9hG4bK-wnelfgn8qinn;rport f: "TestSnom1" <sip:[email protected]>;tag=bj35colkje t: <sip:[email protected];user=phone> i: 3c433d50d289-bfx9jv1z1nk1 CSeq: 1 INVITE Max-Forwards: 70 m: <sip:[email protected]:2052;line=t7xm4yiz>;reg-id=1 P-Key-Flags: keys="3" User-Agent: snom300/7.3.14 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 3600;refresher=uas Min-SE: 90 c: application/sdp l: 372 v=0 o=root 105497526 105497526 IN IP4 192.168.150.158 s=call c=IN IP4 192.168.150.158 t=0 0 m=audio 60184 RTP/AVP 18 9 8 0 3 99 4 101 a=rtpmap:18 g729/8000 a=rtpmap:9 g722/8000 a=rtpmap:8 pcma/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:3 gsm/8000 a=rtpmap:99 g726-32/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv ________________________________ Received from udp:192.168.150.19:5060 at 17/4/2009 02:46:41:898 (288 bytes): SIP/2.0 100 Trying From: "TestSnom1" <sip:[email protected]>;tag=bj35colkje To: <sip:[email protected];user=phone> Call-Id: 3c433d50d289-bfx9jv1z1nk1 Cseq: 1 INVITE Via: SIP/2.0/UDP 192.168.150.158:2052;branch=z9hG4bK-wnelfgn8qinn;rport=2052 Content-Length: 0 ________________________________ Received from udp:192.168.150.19:5060 at 17/4/2009 02:46:42:011 (520 bytes): SIP/2.0 407 Proxy Authentication Required From: "TestSnom1" <sip:[email protected]>;tag=bj35colkje To: <sip:[email protected];user=phone>;tag=8350a339 Call-Id: 3c433d50d289-bfx9jv1z1nk1 Cseq: 1 INVITE Via: SIP/2.0/UDP 192.168.150.158:2052;branch=z9hG4bK-wnelfgn8qinn;rport=2052 Proxy-Authenticate: Digest realm="sipxecs.domain.test", nonce="4cedb83120a2e56bfce3f20c6a41ce3349e87a31" Server: sipXecs/4.1.0 sipXecs/sipXproxy (Linux) Date: Fri, 17 Apr 2009 12:46:41 GMT Content-Length: 0 ________________________________ Sent to udp:192.168.150.19:5060 at 17/4/2009 02:46:42:017 (379 bytes): ACK sip:[email protected];user=phone SIP/2.0 v: SIP/2.0/UDP 192.168.150.158:2052;branch=z9hG4bK-wnelfgn8qinn;rport f: "TestSnom1" <sip:[email protected]>;tag=bj35colkje t: <sip:[email protected];user=phone>;tag=8350a339 i: 3c433d50d289-bfx9jv1z1nk1 CSeq: 1 ACK Max-Forwards: 70 m: <sip:[email protected]:2052;line=t7xm4yiz>;reg-id=1 l: 0 ________________________________ Sent to udp:192.168.150.19:5060 at 17/4/2009 02:46:42:032 (1306 bytes): INVITE sip:[email protected];user=phone SIP/2.0 v: SIP/2.0/UDP 192.168.150.158:2052;branch=z9hG4bK-q2d9nldocwrl;rport f: "TestSnom1" <sip:[email protected]>;tag=bj35colkje t: <sip:[email protected];user=phone> i: 3c433d50d289-bfx9jv1z1nk1 CSeq: 2 INVITE Max-Forwards: 70 m: <sip:[email protected]:2052;line=t7xm4yiz>;reg-id=1 P-Key-Flags: keys="3" User-Agent: snom300/7.3.14 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 3600;refresher=uas Min-SE: 90 Proxy-Authorization: Digest username="974",realm="sipxecs.domain.test",nonce="4cedb83120a2e56bfce3f20c6a41ce3349e87a31",uri="sip:[email protected];user=phone",response="78c0b7444abc7c39436512a6d259e3d6",algorithm=MD5 c: application/sdp l: 372 v=0 o=root 105497526 105497526 IN IP4 192.168.150.158 s=call c=IN IP4 192.168.150.158 t=0 0 m=audio 60184 RTP/AVP 18 9 8 0 3 99 4 101 a=rtpmap:18 g729/8000 a=rtpmap:9 g722/8000 a=rtpmap:8 pcma/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:3 gsm/8000 a=rtpmap:99 g726-32/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv ________________________________ Received from udp:192.168.150.19:5060 at 17/4/2009 02:46:42:040 (520 bytes): SIP/2.0 407 Proxy Authentication Required From: "TestSnom1" <sip:[email protected]>;tag=bj35colkje To: <sip:[email protected];user=phone>;tag=8350a339 Call-Id: 3c433d50d289-bfx9jv1z1nk1 Cseq: 1 INVITE Via: SIP/2.0/UDP 192.168.150.158:2052;branch=z9hG4bK-wnelfgn8qinn;rport=2052 Proxy-Authenticate: Digest realm="sipxecs.domain.test", nonce="4cedb83120a2e56bfce3f20c6a41ce3349e87a31" Server: sipXecs/4.1.0 sipXecs/sipXproxy (Linux) Date: Fri, 17 Apr 2009 12:46:41 GMT Content-Length: 0 ________________________________ Sent to udp:192.168.150.19:5060 at 17/4/2009 02:46:42:041 (379 bytes): ACK sip:[email protected];user=phone SIP/2.0 v: SIP/2.0/UDP 192.168.150.158:2052;branch=z9hG4bK-wnelfgn8qinn;rport f: "TestSnom1" <sip:[email protected]>;tag=bj35colkje t: <sip:[email protected];user=phone>;tag=8350a339 i: 3c433d50d289-bfx9jv1z1nk1 CSeq: 1 ACK Max-Forwards: 70 m: <sip:[email protected]:2052;line=t7xm4yiz>;reg-id=1 l: 0 ________________________________ Received from udp:192.168.150.19:5060 at 17/4/2009 02:46:42:102 (288 bytes): SIP/2.0 100 Trying From: "TestSnom1" <sip:[email protected]>;tag=bj35colkje To: <sip:[email protected];user=phone> Call-Id: 3c433d50d289-bfx9jv1z1nk1 Cseq: 2 INVITE Via: SIP/2.0/UDP 192.168.150.158:2052;branch=z9hG4bK-q2d9nldocwrl;rport=2052 Content-Length: 0 ________________________________ Received from udp:192.168.150.19:5060 at 17/4/2009 02:46:42:203 (720 bytes): SIP/2.0 180 Ringing From: "TestSnom1" <sip:[email protected]>;tag=bj35colkje To: <sip:[email protected];user=phone>;tag=624039693 Call-Id: 3c433d50d289-bfx9jv1z1nk1 Cseq: 2 INVITE Via: SIP/2.0/UDP 192.168.150.158:2052;branch=z9hG4bK-q2d9nldocwrl;rport=2052 Record-Route: <sip:192.168.150.19:5060;lr;sipXecs-rs=%2Aauth%7E.%2Afrom%7EYmozNWNvbGtqZQ%60%60%218caf4b3cc5949503bb7825724ff5eb43> Contact: <sip:[email protected]:5150;LINEID=9e076396a5c2c2ad9c54ca3c29554a6d> Date: Fri, 17 Apr 2009 12:46:42 GMT User-Agent: sipXacd (4.1.0-015201) (Linux) Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, INFO, REGISTER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 ________________________________ Received from udp:192.168.150.19:5060 at 17/4/2009 02:46:42:297 (985 bytes): SIP/2.0 200 OK From: "TestSnom1" <sip:[email protected]>;tag=bj35colkje To: <sip:[email protected];user=phone>;tag=624039693 Call-Id: 3c433d50d289-bfx9jv1z1nk1 Cseq: 2 INVITE Contact: <sip:[email protected]:5150;LINEID=9e076396a5c2c2ad9c54ca3c29554a6d> Content-Type: application/sdp Content-Length: 214 Via: SIP/2.0/UDP 192.168.150.158:2052;branch=z9hG4bK-q2d9nldocwrl;rport=2052 Record-Route: <sip:192.168.150.19:5060;lr;sipXecs-rs=%2Aauth%7E.%2Afrom%7EYmozNWNvbGtqZQ%60%60%218caf4b3cc5949503bb7825724ff5eb43> Session-Expires: 3600 Date: Fri, 17 Apr 2009 12:46:42 GMT Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, INFO, REGISTER, SUBSCRIBE, NOTIFY User-Agent: sipXacd (4.1.0-015201) (Linux) Accept-Language: en Supported: replaces v=0 o=sipXecs 5 5 IN IP4 192.168.150.19 s=phone-call c=IN IP4 192.168.150.19 t=0 0 m=audio 10000 RTP/AVP 8 0 101 a=sendrecv a=rtpmap:8 pcma/8000/1 a=rtpmap:0 pcmu/8000/1 a=rtpmap:101 telephone-event/8000/1 ________________________________ Sent to udp:192.168.150.19:5060 at 17/4/2009 02:46:42:307 (528 bytes): ACK sip:[email protected]:5150;LINEID=9e076396a5c2c2ad9c54ca3c29554a6d SIP/2.0 v: SIP/2.0/UDP 192.168.150.158:2052;branch=z9hG4bK-4zetnkpayupe;rport Route: <sip:192.168.150.19:5060;lr;sipXecs-rs=%2Aauth%7E.%2Afrom%7EYmozNWNvbGtqZQ%60%60%218caf4b3cc5949503bb7825724ff5eb43> f: "TestSnom1" <sip:[email protected]>;tag=bj35colkje t: <sip:[email protected];user=phone>;tag=624039693 i: 3c433d50d289-bfx9jv1z1nk1 CSeq: 2 ACK Max-Forwards: 70 m: <sip:[email protected]:2052;line=t7xm4yiz>;reg-id=1 l: 0 ________________________________ Received from udp:192.168.150.19:5060 at 17/4/2009 02:46:42:425 (985 bytes): SIP/2.0 200 OK From: "TestSnom1" <sip:[email protected]>;tag=bj35colkje To: <sip:[email protected];user=phone>;tag=624039693 Call-Id: 3c433d50d289-bfx9jv1z1nk1 Cseq: 2 INVITE Contact: <sip:[email protected]:5150;LINEID=9e076396a5c2c2ad9c54ca3c29554a6d> Content-Type: application/sdp Content-Length: 214 Via: SIP/2.0/UDP 192.168.150.158:2052;branch=z9hG4bK-q2d9nldocwrl;rport=2052 Record-Route: <sip:192.168.150.19:5060;lr;sipXecs-rs=%2Aauth%7E.%2Afrom%7EYmozNWNvbGtqZQ%60%60%218caf4b3cc5949503bb7825724ff5eb43> Session-Expires: 3600 Date: Fri, 17 Apr 2009 12:46:42 GMT Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, INFO, REGISTER, SUBSCRIBE, NOTIFY User-Agent: sipXacd (4.1.0-015201) (Linux) Accept-Language: en Supported: replaces v=0 o=sipXecs 5 5 IN IP4 192.168.150.19 s=phone-call c=IN IP4 192.168.150.19 t=0 0 m=audio 10000 RTP/AVP 8 0 101 a=sendrecv a=rtpmap:8 pcma/8000/1 a=rtpmap:0 pcmu/8000/1 a=rtpmap:101 telephone-event/8000/1 ________________________________ Sent to udp:192.168.150.19:5060 at 17/4/2009 02:46:42:427 (528 bytes): ACK sip:[email protected]:5150;LINEID=9e076396a5c2c2ad9c54ca3c29554a6d SIP/2.0 v: SIP/2.0/UDP 192.168.150.158:2052;branch=z9hG4bK-4zetnkpayupe;rport Route: <sip:192.168.150.19:5060;lr;sipXecs-rs=%2Aauth%7E.%2Afrom%7EYmozNWNvbGtqZQ%60%60%218caf4b3cc5949503bb7825724ff5eb43> f: "TestSnom1" <sip:[email protected]>;tag=bj35colkje t: <sip:[email protected];user=phone>;tag=624039693 i: 3c433d50d289-bfx9jv1z1nk1 CSeq: 2 ACK Max-Forwards: 70 m: <sip:[email protected]:2052;line=t7xm4yiz>;reg-id=1 l: 0 ________________________________ Received from udp:192.168.150.19:5060 at 17/4/2009 02:46:48:834 (692 bytes): BYE sip:[email protected]:2052;line=t7xm4yiz;x-sipX-nonat SIP/2.0 From: <sip:[email protected];user=phone>;tag=624039693 To: "TestSnom1" <sip:[email protected]>;tag=bj35colkje Call-Id: 3c433d50d289-bfx9jv1z1nk1 Cseq: 1 BYE Contact: <sip:[email protected]:5150;LINEID=9e076396a5c2c2ad9c54ca3c29554a6d;x-sipX-nonat> Date: Fri, 17 Apr 2009 12:46:48 GMT Max-Forwards: 19 User-Agent: sipXacd (4.1.0-015201) (Linux) Accept-Language: en Supported: replaces Via: SIP/2.0/UDP 192.168.150.19;branch=z9hG4bK-sipXecs-00836321a900fb0fc453a9b44c18ad50cc4b Via: SIP/2.0/UDP 192.168.150.19:5150;branch=z9hG4bK-sipXecs-00be1158fe64df32a8248079ea542964915e Content-Length: 0 ________________________________ Sent to udp:192.168.150.19:5060 at 17/4/2009 02:46:48:880 (607 bytes): SIP/2.0 200 OK v: SIP/2.0/UDP 192.168.150.19;branch=z9hG4bK-sipXecs-00836321a900fb0fc453a9b44c18ad50cc4b v: SIP/2.0/UDP 192.168.150.19:5150;branch=z9hG4bK-sipXecs-00be1158fe64df32a8248079ea542964915e f: <sip:[email protected];user=phone>;tag=624039693 t: "TestSnom1" <sip:[email protected]>;tag=bj35colkje i: 3c433d50d289-bfx9jv1z1nk1 CSeq: 1 BYE m: <sip:[email protected]:2052;line=t7xm4yiz>;reg-id=1 User-Agent: snom300/7.3.14 RTP-RxStat: Total_Rx_Pkts=320,Rx_Pkts=320,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0 RTP-TxStat: Total_Tx_Pkts=330,Tx_Pkts=330,Remote_Tx_Pkts=319 l: 0 ________________________________ Am I missing something, is it a coding problem or why do I always here that noise when calling a hotline (doesn't matter if it's an internal call or it comes from external) ? Help would be really appreciated! Best regards Stefan
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