[Freeswitch-users] SRTP disabling
Steven Ayre
steveayre at gmail.com
Wed Feb 13 20:59:23 MSK 2013
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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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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