[Freeswitch-users] SRTP disabling

Wed Feb 13 20:51:08 MSK 2013

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 at 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 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
>
>
>
>
>   _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 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.
>
>
>
>
> _lvnd_
>  {^_^}
>
>
>
>
>
>
> 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 ?
>
>
>
>
> _lvnd_
>  {^_^}
>
>
>
>
>
>
> 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
>
>
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