In normal situations, I will do so. But right now I don't have chance to wait for them to make a fix. Besides, I already made the necessary fix on FS. I changed sdp parser. It ignores a=crypto attributes and set media protocol as RTP/AVP even if coming SDP offer is RTP/SAVP. I bind this behaviour to a new proprieratary channel attribute. It works just fine according to my smoke tests. -- _lvnd_ {^_^} On Thu, 2013-02-14 at 13:41 -0500, Michael Jerris wrote: Why not just ask yealink to fix this? They have been pretty good about making fixes for us. Mike On Feb 14, 2013, at 10:03 AM, Levend Sayar <levend.sayar@karel.com.tr<mailto:levend.sayar@karel.com.tr>> wrote: Thanks for your kind responses Steve. Phones are same brand, you are right. And not Polycom or Linksys. They are Yealink phones. I found the code piece handling the RTP/SAVP part you mentioned. (sofia_glue_negotiate_sdp function). Although it is not wise, I will add a new channel variable and try to disable SRTP even if phone says SRTP is mandatory. I need that. On Wed, 2013-02-13 at 17:59 +0000, Steven Ayre wrote: If the phone is Polycom or Linksys have you read the notes on http://wiki.freeswitch.org/wiki/SRTP? On 13 February 2013 17:53, Steven Ayre <steveayre@gmail.com<mailto:steveayre@gmail.com>> wrote: Nevertheless it's not being negotiated and this is possibly a bug in the phone. Adding a workaround in FS would decrease security because the phone is explicitly saying it'll only accept SRTP so sending plain RTP wouldn't be wise. I assume both phones are the same make? On 7 February 2013 13:54, Levend Sayar <levend.sayar@karel.com.tr<mailto:levend.sayar@karel.com.tr>> wrote: But the very same phone calls another phone and talk with RTP, not SRTP if the peer does not accept SRTP Here is the SDP offer by the same phone v=0 o=- 20186 20186<tel:20186%2020186> IN IP4 192.168.173.69 s=SDP data c=IN IP4 192.168.173.69 t=0 0 m=audio 11782 RTP/SAVP 0 8 18 9 101 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:NzFmYjdiMjk1OTY2ODQwYzExZjM0ZmE2NGM0YWMw a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:M2MxMTE2OWFjOGY2ZjEwADEzZmZkNzAxNjRlMzFm a=crypto:3 F8_128_HMAC_SHA1_80 inline:NjkzZDg2Mjk0ZTkxMjg1YzdmYjFiNjRlMmFhNGFm a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:9 G722/8000 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=ptime:20 a=sendrecv And here is the SDP answer sent by the other phone v=0 o=- 20029 20029 IN IP4 192.168.173.65 s=SDP data c=IN IP4 192.168.173.65 t=0 0 m=audio 11794 RTP/SAVP 0 101 a=rtpmap:0 PCMU/8000 a=sendrecv a=ptime:20 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 On Thu, 2013-02-07 at 13:37 +0000, Steven Ayre wrote: m=audio 11780 RTP/SAVP 0 8 18 9 101 RTP/SAVP means SRTP is mandatory. You need to reconfigure the phone. If the phone sends RTP/AVP then that means plain RTP, and RTP/AVP with a a=crypto attribute means SRTP is optional. On 7 February 2013 13:26, Levend Sayar <levend.sayar@karel.com.tr<mailto:levend.sayar@karel.com.tr>> wrote: Below is the SDP offer sent by the phone. v=0 o=- 20185 20185<tel:20185%2020185> IN IP4 192.168.173.69 s=SDP data c=IN IP4 192.168.173.69 t=0 0 m=audio 11780 RTP/SAVP 0 8 18 9 101 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:MTM2MjVhMGI1NDZjYmRjADU5NWVjNGVkNTNlYzA1 a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:YmExYmZhNQAzN2ZjNDgzYTRkNGU2ZjFiN2Q0MmE3 a=crypto:3 F8_128_HMAC_SHA1_80 inline:N2Q2NTRiYQAxZjA3MWY3ZjI1YTI5NjIyM2FjODYw a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:9 G722/8000 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=ptime:20 a=sendrecv And below is the SDP answer sent by FS v=0 o=FreeSWITCH 1360230601 1360230602 IN IP4 192.168.169.114 s=FreeSWITCH c=IN IP4 192.168.169.114 t=0 0 m=audio 12532 RTP/SAVP 9 101 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:l8v0R64H7CP0vEx9j0Ycdbob8bgMCpLDppWGy7Dy On Thu, 2013-02-07 at 13:09 +0000, Steven Ayre wrote: What I mean is you'll see two separate m=audio lines within the callee's SDP, one for 'RTP/AVP' and one for 'SRTP/AVP'. If there is no m=audio line for RTP/AVP the caller won't know of a port that's expecting RTP. So if the callee only sends SRTP/AVP the caller can't send RTP. Can you show us the SDP being sent by the phone? On 7 February 2013 11:01, Levend Sayar <levend.sayar@karel.com.tr<mailto:levend.sayar@karel.com.tr>> wrote: Thanx Steven. Caller makes the offer for SDP but callee chooses whatever it wants. So caller can offer SRTP but callee can prefer not to talk encrypted. In our case I want FS to choose non secure media. Phone will offer SRTP on the conference call but FS must prefer RTP, not SRTP. On 7 Şub 2013, at 11:13, "Steven Ayre" <steveayre@gmail.com<mailto:steveayre@gmail.com>> wrote: It's also going to rely on the phone actually offering RTP/AVP as well as SRTP/AVP in their SDP - without that there'd be nowhere to send insecure RTP. On 6 February 2013 16:09, Levend Sayar <levend.sayar@karel.com.tr<mailto:levend.sayar@karel.com.tr>> wrote: Thanks Daniel for the reply. I tried <action application="set" data="sip_secure_media=false" /> But did not work. Upon your reply I also tried <action application="set" data="secure_media=false" /> But did not work either. I am doing something wrong ? On 6 Şub 2013, at 18:00, "Daniel Ivanov" <sertys@gmail.com<mailto:sertys@gmail.com>> wrote: Of course you can. Just set the secure_media var to false and you will be srtp-free in sip. On Feb 5, 2013 6:06 PM, "Levend Sayar" <levend.sayar@karel.com.tr<mailto:levend.sayar@karel.com.tr>> wrote: Hi all. I am using FS as a conference server. Some of my phones are using SRTP , some of them not. Both type of phone can join a conference. FS can talk to each peer with SRTP or not depending on the phone itself. My question: Is it possible to disable SRTP on FS ? I suppose if i can disable SRTP, FS will talk without SRTP with each phone whether they are using SRTP or not.