intParse (Medisys) Posted June 25, 2007 Report Share Posted June 25, 2007 Facing a serious problem with Snom360 phones about the white noise being generated on calls after 3-4 minutes (sometimes shorter; 30sec) on an srtp connection. I have tested and seen the same symptoms for srtp calls on both linux(SuSe) and windows (2000) machines. PBXnSIP v18.104.22.168 is running on Suse machine and v22.214.171.124 is running on windows2000 machine. All phones were upgraded to Snom360 6.5.10. I even upgraded them to snom360 7.1.9 version but couldn't get the phones make secure (srtp) calls on v7. (BTW v7 upgrade slows down the snom360 phones. Startup and registration was extremely slow. Keep away from v7 if you own snom360 phones.) I have also tried the sip:xxx.xxx.xxx.xxx:5061;transport=tls line on the outgoing proxy and no luck. In the PBXnSIP 1.5 release notes some of the srtp problems were mentioned as being solved but I don't know if they apply to this white noise situation. I've read some of the threads mentioning this problem but couldn't find any solution to this problem. I am listing the threads I found in PBXnSIP and others forums below (namely asterisk forum thread about the snom white noise); http://forum.pbxnsip.com/index.php?showtopic=93 <-- (Check the last answer or do a text search on "white noise) http://bugs.digium.com/print_bug_page.php?bug_id=5413 <--- (do a text search on "white noise") on the second thread it says "When chan_sip is starting a non crypto call, sending out an INVITE without any sdesc (_SIP_SRTP_SDES is unset), but the reply from the peer includes a sdes proposal, a crypto policy get set up and activated, without the local key being exchanged. At least Snom 5.x always answer INVITE with a a=crypto line and "a=encryption:optional" in the sdp, calling one without setting _SIP_SRTP_SDES gives asterisk using srtp and the snom expecting unenctypted rtp, and nice white noise in the earpiece." So here are my questions: * Is this white noise being generated because of snom neglecting to add the a=crypto line on some of the sip messages while sending them to PBXnSIP? * If it is, is there a work around for this? (w/o upgrading to PBXnSIP 2.0) Any help greatly appreciated, thanks in advance. Quote Link to comment Share on other sites More sharing options...
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