[Freeswitch-users] SRTP disabling

Thu Feb 14 18:03:51 MSK 2013

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 at gmail.com<mailto: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<mailto: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<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 at karel.com.tr<mailto:levend.sayar at 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 at karel.com.tr<mailto: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<mailto: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<mailto: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<mailto: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<mailto: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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