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. TIA -- _lvnd_ {^_^}
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> 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.
TIA
--
_lvnd_ {^_^}
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
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 ? İyi çalışmalar. _lvnd_ {^_^} KAREL`UCAP 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. TIA -- _lvnd_ {^_^} _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto: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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto: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
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. -Steve On 6 February 2013 16:09, Levend Sayar <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 ?
İyi çalışmalar.
_lvnd_ {^_^}
KAREL`UCAP
On 6 Şub 2013, at 18:00, "Daniel Ivanov" <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> 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.
TIA
--
_lvnd_ {^_^}
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
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. İyi çalışmalar. _lvnd_ {^_^} KAREL`UCAP 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. -Steve 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 ? İyi çalışmalar. _lvnd_ {^_^} KAREL`UCAP 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. TIA -- _lvnd_ {^_^} _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto: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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto: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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto: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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org
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? -Steve On 7 February 2013 11:01, Levend Sayar <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.
İyi çalışmalar.
_lvnd_ {^_^}
KAREL`UCAP
On 7 Şub 2013, at 11:13, "Steven Ayre" <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.
-Steve
On 6 February 2013 16:09, Levend Sayar <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 ?
İyi çalışmalar.
_lvnd_ {^_^}
KAREL`UCAP
On 6 Şub 2013, at 18:00, "Daniel Ivanov" <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> 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.
TIA
--
_lvnd_ {^_^}
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
Below is the SDP offer sent by the phone. v=0 o=- 20185 20185 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 -- İyi çalışmalar. _lvnd_ {^_^} KAREL`UCAP 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? -Steve 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. İyi çalışmalar. _lvnd_ {^_^} KAREL`UCAP 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. -Steve 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 ? İyi çalışmalar. _lvnd_ {^_^} KAREL`UCAP 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. TIA -- _lvnd_ {^_^} _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto: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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto: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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto: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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto: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 plain text document attachment (ATT00001) _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto: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
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. -Steve On 7 February 2013 13:26, Levend Sayar <levend.sayar@karel.com.tr> wrote:
** Below is the SDP offer sent by the phone.
v=0 o=- 20185 20185 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
-- İyi çalışmalar.
_lvnd_ {^_^}
KAREL`UCAP
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?
-Steve
On 7 February 2013 11:01, Levend Sayar <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.
İyi çalışmalar.
_lvnd_
{^_^}
KAREL`UCAP
On 7 Şub 2013, at 11:13, "Steven Ayre" <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.
-Steve
On 6 February 2013 16:09, Levend Sayar <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 ?
İyi çalışmalar.
_lvnd_
{^_^}
KAREL`UCAP
On 6 Şub 2013, at 18:00, "Daniel Ivanov" <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> 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.
TIA
--
_lvnd_ {^_^}
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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 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 -- İyi çalışmalar. _lvnd_ {^_^} KAREL`UCAP 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. -Steve 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 -- İyi çalışmalar. _lvnd_ {^_^} KAREL`UCAP 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? -Steve 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. İyi çalışmalar. _lvnd_ {^_^} KAREL`UCAP 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. -Steve 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 ? İyi çalışmalar. _lvnd_ {^_^} KAREL`UCAP 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. TIA -- _lvnd_ {^_^} _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto: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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto: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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 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Hi again. I checked the FS code and i see that rpm i am using is built with ZRTP disabled. There is only one place that "sip_secure_media" on the code. That is on switch_rtp.c #ifdef ENABLE_ZRTP if (zrtp_on) { switch_rtp_t *master_rtp_session = NULL; int initiator = 0; const char *zrtp_enabled = switch_channel_get_variable(channel, "zrtp_secure_media"); const char *srtp_enabled = switch_channel_get_variable(channel, "sip_secure_media"); So since ENABLE_ZRTP is 0, i don't have chance to use "sip_secure_media" variable. Is there any other variable that i can use and make sofia module not to choose SRTP ? _lvnd_ {^_^} On Thu, 2013-02-07 at 13:54 +0000, Levend Sayar 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 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 -- _lvnd_ {^_^} 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. -Steve 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 _lvnd_ {^_^} 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? -Steve 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. _lvnd_ {^_^} 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. -Steve 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 ? _lvnd_ {^_^} 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. TIA plain text document attachment (ATT00001) _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto: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
sip_secure_media also covers SRTP without ZRTP. It's actually used elsewhere too, but aliased. src/mod/endpoints/mod_sofia/mod_sofia.h:115:#define SOFIA_SECURE_MEDIA_VARIABLE "sip_secure_media" Search the source tree for SOFIA_SECURE_MEDIA_VARIABLE and you'll see it's used in a number of other places too. -Steve On 13 February 2013 17:13, Levend Sayar <levend.sayar@karel.com.tr> wrote:
** Hi again.
I checked the FS code and i see that rpm i am using is built with ZRTP disabled. There is only one place that "sip_secure_media" on the code. That is on switch_rtp.c
#ifdef ENABLE_ZRTP if (zrtp_on) { switch_rtp_t *master_rtp_session = NULL
int initiator = 0; const char *zrtp_enabled = switch_channel_get_variable(channel, "zrtp_secure_media"); const char *srtp_enabled = switch_channel_get_variable(channel, "sip_secure_media");
So since ENABLE_ZRTP is 0, i don't have chance to use "sip_secure_media" variable.
Is there any other variable that i can use and make sofia module not to choose SRTP ?
_lvnd_ {^_^}
On Thu, 2013-02-07 at 13:54 +0000, Levend Sayar 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 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
--
_lvnd_ {^_^}
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.
-Steve
On 7 February 2013 13:26, Levend Sayar <levend.sayar@karel.com.tr> wrote:
Below is the SDP offer sent by the phone.
v=0 o=- 20185 20185 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
_lvnd_ {^_^}
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?
-Steve
On 7 February 2013 11:01, Levend Sayar <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.
_lvnd_ {^_^}
On 7 Şub 2013, at 11:13, "Steven Ayre" <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.
-Steve
On 6 February 2013 16:09, Levend Sayar <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 ?
_lvnd_ {^_^}
On 6 Şub 2013, at 18:00, "Daniel Ivanov" <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> 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.
TIA
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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? -Steve On 7 February 2013 13:54, Levend Sayar <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 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
-- İyi çalışmalar.
_lvnd_ {^_^}
KAREL`UCAP
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.
-Steve
On 7 February 2013 13:26, Levend Sayar <levend.sayar@karel.com.tr> wrote:
Below is the SDP offer sent by the phone.
v=0 o=- 20185 20185 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
-- İyi çalışmalar.
_lvnd_ {^_^}
KAREL`UCAP
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?
-Steve
On 7 February 2013 11:01, Levend Sayar <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.
İyi çalışmalar.
_lvnd_ {^_^}
KAREL`UCAP
On 7 Şub 2013, at 11:13, "Steven Ayre" <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.
-Steve
On 6 February 2013 16:09, Levend Sayar <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 ?
İyi çalışmalar.
_lvnd_ {^_^}
KAREL`UCAP
On 6 Şub 2013, at 18:00, "Daniel Ivanov" <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> 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.
TIA
--
_lvnd_ {^_^}
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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If the phone is Polycom or Linksys have you read the notes on http://wiki.freeswitch.org/wiki/SRTP? -Steve On 13 February 2013 17:53, Steven Ayre <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?
-Steve
On 7 February 2013 13:54, Levend Sayar <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 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
-- İyi çalışmalar.
_lvnd_ {^_^}
KAREL`UCAP
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.
-Steve
On 7 February 2013 13:26, Levend Sayar <levend.sayar@karel.com.tr> wrote:
Below is the SDP offer sent by the phone.
v=0 o=- 20185 20185 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
-- İyi çalışmalar.
_lvnd_ {^_^}
KAREL`UCAP
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?
-Steve
On 7 February 2013 11:01, Levend Sayar <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.
İyi çalışmalar.
_lvnd_ {^_^}
KAREL`UCAP
On 7 Şub 2013, at 11:13, "Steven Ayre" <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.
-Steve
On 6 February 2013 16:09, Levend Sayar <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 ?
İyi çalışmalar.
_lvnd_ {^_^}
KAREL`UCAP
On 6 Şub 2013, at 18:00, "Daniel Ivanov" <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> 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.
TIA
--
_lvnd_ {^_^}
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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. -- _lvnd_ {^_^} 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? -Steve 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? -Steve 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 -- İyi çalışmalar. _lvnd_ {^_^} KAREL`UCAP 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. -Steve 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 -- İyi çalışmalar. _lvnd_ {^_^} KAREL`UCAP 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? -Steve 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. İyi çalışmalar. _lvnd_ {^_^} KAREL`UCAP 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. -Steve 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 ? İyi çalışmalar. _lvnd_ {^_^} KAREL`UCAP 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. TIA -- _lvnd_ {^_^} _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto: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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official 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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> 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> 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> 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 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> wrote: Below is the SDP offer sent by the phone.
v=0 o=- 20185 20185 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> 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> 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> 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> 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> 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.
hello guys, I need to find the ip address of a sip phone registered to a freeswitch server any body knows how? thanks in advance regards, Sherif
From the fs command line
sofia status profile PROFILE_NAME reg
That will show you the registrations active, and you should be able to see the IP in the contact field. Nick On Thu, Feb 14, 2013 at 10:55 AM, Sherif Omran <sherifomran2000@yahoo.com>wrote:
hello guys,
I need to find the ip address of a sip phone registered to a freeswitch server
any body knows how?
thanks in advance
regards, Sherif
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If your phone is behind NAT, the Call-ID (from the command Nick provided you) will also give you the internal IP which is kinda neat. Also, you can include the phone number after the reg to only show registration for this particular phone:
sofia status profile PROFILE_NAME reg PHONE_NUMBER
Benoit De : freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] De la part de Nick Vines Envoyé : 14 février 2013 14:08 À : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] find sip phone ip
From the fs command line
sofia status profile PROFILE_NAME reg
That will show you the registrations active, and you should be able to see the IP in the contact field. Nick On Thu, Feb 14, 2013 at 10:55 AM, Sherif Omran <sherifomran2000@yahoo.com> wrote: hello guys, I need to find the ip address of a sip phone registered to a freeswitch server any body knows how? thanks in advance regards, Sherif _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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
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.
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> 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. TIA -- _lvnd_ {^_^} _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com 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
participants (7)
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Benoit Raymond
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Daniel Ivanov
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Levend Sayar
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Michael Jerris
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Nick Vines
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Sherif Omran
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Steven Ayre