srtp-users — list for libsrtp users

[Srtp-users] SRTCP encryption

[Julien C. <jch...@gm...>]

Looking at :
https://tools.ietf.org/html/rfc3711

"  from the ninth (9) octet to the end of the compound packet. "

SRCTP is encrypting everything besides the 8 octets header.
This is also what I found in the source code ...

However, I also saw that WebRTC was using libsrtp and I was wondering why I
was able to read the SR and RR blocks. (I know this is probably not the
right list to ask about WebRTC specific question).

Since SRTP does not encrypt sequence id and timestamp, I was expecting the
SR and RR blocks would also be left in clear.

Am I right to conclude that there should be only 2 options :

1- encrypt everything after the header (first 8 octets)
2- encrypt nothing

Appreciated
Julien

[Srtp-users] A beginner's question

[Patrick R. <int...@ya...>]

Hello List ;-)

Am programming an RTP relay whose job is to forward RTP frames from an in-house source to a browser running a javascript based html5 player. So far I have dealt with libnice for the STUN connection, openssl for the handshake and now libsrtp2 because apparently browsers want their RTP encrypted.

Now Chrome is telling me:

ERROR:srtptransport.cc(210)] Failed to unprotect RTP packet: size=1474, seqnum=6846, SSRC=1908664075 for each frame I send

so my understanding is that somehow I fail to protect the packet correctly.

sequence of events:

SRTP_PROTECTION_PROFILE *srtp_profile = SSL_get_selected_srtp_profile(ssl);


switch (srtp_profile->id) {
case SRTP_AES128_CM_SHA1_80:
case SRTP_AES128_CM_SHA1_32:
key_length = SRTP_MASTER_KEY_LENGTH;
salt_length = SRTP_MASTER_SALT_LENGTH;
master_length = SRTP_MASTER_LENGTH;
break;

default:
/* Will never happen? */
break;
}





/* Export keying material for SRTP */
SSL_export_keying_material(ssl, key, master_length * 2, "EXTRACTOR-dtls_srtp", 19, NULL, 0, 0);

/* Key derivation (http://tools.ietf.org/html/rfc5764#section-4.2) */

local_key = key;
remote_key = local_key + key_length;
local_salt = remote_key + key_length;
remote_salt = local_salt + salt_length;

/* Build master keys and set SRTP policies */

switch (srtp_profile->id) {
case SRTP_AES128_CM_SHA1_80:
srtp_crypto_policy_set_aes_cm_128_hmac_sha1_80(&policy.rtp);
srtp_crypto_policy_set_aes_cm_128_hmac_sha1_80(&policy.rtcp);
break;
case SRTP_AES128_CM_SHA1_32:
srtp_crypto_policy_set_aes_cm_128_hmac_sha1_32(&policy.rtp);
srtp_crypto_policy_set_aes_cm_128_hmac_sha1_80(&policy.rtcp);
break;
default:
/* Will never happen? */
break;
}


then:


error = srtp_init();
ssrc.type = ssrc_any_outbound;
ssrc.value = randomssrc;
policy.ssrc = ssrc;
policy.key = (unsigned char *) local_key;
memcpy(policy.key, local_key, key_length);
memcpy(policy.key + key_length, local_salt, salt_length);
policy.next = NULL;
// allocate and initialize the SRTP session
error = srtp_create(&session, &policy);


and finally:
(len and buf _packet are extracted from a circular buffer prior to these)

int length = len - 12;
srtp_protect_rtcp(session, buf_packet, &length);
memcpy(packet_to_send + 12, buf_packet, length);
ret = nice_agent_send(agent, stream_id, 1, length + 12, packet_to_send);


So maybe someone can help to correct that sequence of events, as I said this is my first attempt at building anything that has to do with any of the above ...

Any help appreciated.

Thanks

/Patrick

[Srtp-users] Fwd: SRTP / SRTCP Key

[Austin E. <aus...@gm...>]

Anybody can answer my query please ?


---------- Forwarded message ----------
From: Austin Einter <aus...@gm...>
Date: Sat, Mar 17, 2018 at 4:23 PM
Subject: SRTP / SRTCP Key
To: srt...@li...


Hi
I am using libSRTP to encrypt RTP traffic now. I do the DTLS handshake
between two endpoints, extract the key and use this key for SRTP.

My question is for RTCP which key I need to use.
I have two options

1) For RTCP (if using separate port) do DTLS handshake and derive the key
and use it.
2) OR use the key already derived in RTP case.

Kindly let me know which option is normally followed 1) or 2).

Thanks
Austin

[Srtp-users] SRTP / SRTCP Key

[Austin E. <aus...@gm...>]

Hi
I am using libSRTP to encrypt RTP traffic now. I do the DTLS handshake
between two endpoints, extract the key and use this key for SRTP.

My question is for RTCP which key I need to use.
I have two options

1) For RTCP (if using separate port) do DTLS handshake and derive the key
and use it.
2) OR use the key already derived in RTP case.

Kindly let me know which option is normally followed 1) or 2).

Thanks
Austin

[Srtp-users] Soliciting feedback on new AES-GCM related bug found in libSRTP 1.5.4 & 2.0.0

[Pascal B. (pabuhler) <pab...@ci...>]

Hi,

The following bug was recently reported on github https://github.com/cisco/libsrtp/issues/256, this bug relates to AES-GCM on little-endian system.
Fixing this bug will cause some backward compatibility issues with existing libSRTP version, but not fixing is not really an option due to interop issues with other AES-GCM implementations as well as it being a potential security problem.
Current plan is to fix as part of the upcoming 2.1.0 release and then simultaneously create patch release for 1.5.x.
If this bug affects your usage of libSRTP and you would like to discuss how a fix is released then please reply to this mail at li...@li...<mailto:li...@li...> or else comment directly on the github issue.

pascal

[Srtp-users] libSRTP project has moved, this mailer is retired!

[Pascal B. (pabuhler) <pab...@ci...>]

Hi,

For those that do not know already the libSRTP project has completely moved from https://sourceforge.net/projects/srtp  to https://github.com/cisco/libsrtp .

As part of this move the mailer's hosted on sourceforge , srt...@li...<mailto:srt...@li...> & srt...@li...<mailto:srt...@li...> can no long be maintained and so are retiring.

There is a new libsrtp mailer set up at li...@li...<mailto:li...@li...> for general news / announcements / discussions. This is an open list, to subscribe go to https://lists.packetizer.com/mailman/listinfo/libsrtp .



Bets regards



Pascal, on behalf of libSRTP maintenance team.

Re: [Srtp-users] manual

[carl h <hey...@gm...>]

Sorry to include the developers list on this one. Mustn't be a very busy
mailing list. Have not got any
responses to my first inquiry.

I noticed later that I had to generate reference docs which I tried to do.

I get this error that I can't resolve (not familiar with text generation
tools).

(/usr/share/texlive/texmf-dist/tex/latex/psnfss/ot1pcr.fd)
[1{/var/lib/texmf/fonts/map/pdftex/updmap/pdftex.map}]
! Undefined control sequence.
l.82 \clearemptydoublepage

libstrp-2.0.0

Appreciate any help.

thanks
/carl h.



On Wed, Jan 11, 2017 at 4:10 PM, carl h <hey...@gm...> wrote:

> Hi,
>
> I'm new to SRTP and getting up to speed.
>
> In the current version (2.0) the reference manual was deleted from the
> release with the reason being that it was out of date. I just read the
> manual and I am wondering which information is incorrect.
>
> Is there somewhere where I can up to date information...or do I have to
> just read the source?
>
> thx
> /c
>

[Srtp-users] manual

[carl h <hey...@gm...>]

Hi,

I'm new to SRTP and getting up to speed.

In the current version (2.0) the reference manual was deleted from the
release with the reason being that it was out of date. I just read the
manual and I am wondering which information is incorrect.

Is there somewhere where I can up to date information...or do I have to
just read the source?

thx
/c

[Srtp-users] Failed to decrypt packets.

[Palanisamy, S. P. <Sud...@ve...>]

Hello Team

Would you please help me to understand what could cause the issue 'Failed to decrypt packets'. I am seeing following errors.

Failed to decrypt packets. Audio buffer size 23932. std run time error: Could not decrypt packet due to: [replay check failed (index too old)]
Failed to decrypt packets. Audio buffer size 28800. std run time error: Encryption flag invalid.  File corruption detected.  Cannot decrypt file.

Thanks,
With warm regards,
-Sudhagar

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Re: [Srtp-users] Stronger ciphers for Asterisk

[John F. <fo...@ci...>]

libsrtp does support those cipher suites.  Having never looked at the 
Asterisk code, I can't comment on the difficultly of enabling these 
cipher suites in Asterisk.  In theory, your proposal looks reasonable.  
I did a similar exercise in PJSIP a few years ago, which was a fairly 
trivial change to PJSIP.  If you're interested in seeing the changes, 
the forked code is located here:

https://github.com/jfigus/PJSIP


On 05/31/2016 02:10 PM, Kevin Long wrote:
>
> Hi folks! First post to the list.
>
>
> I would like to get more (stronger) cipher suites for SRTP allowed in Asterisk.  I am not really C programmer however, and have never submitted a patch to Asterisk.
>
>   It seems that many modern SIP endpoints are supporting more ciphers. In my case I am using a SIP app called Groundwire, which can use
> all the following:
>
> AES_CM_128_HMAC_SHA1_32
> AES_CM_128_HMAC_SHA1_80
> AES_CM_192_HMAC_SHA1_32
> AES_CM_192_HMAC_SHA1_80
> AES_CM_256_HMAC_SHA1_32
> AES_CM_256_HMAC_SHA1_80
> AEAD_AES_128_GCM
> AEAD_AES_256_GCM
>
>
> Asterisk apparently allows only 2  cipher suites for SRTP , both 128 bit .
>
> Below is a snippet from the latest Asterisk stable release tarball - I believe this code section has not been updated in quite some time.   My question is .. could a patch to Asterisk potentially be as simple as just changing this switch/case statement for more options? Or does libSRTP even support the stronger cipher suites listed above?
>
>
>
>
>
>
>
>
>
> static int policy_set_suite(crypto_policy_t *p, enum ast_srtp_suite suite)
> {
>         switch (suite) {
>         case AST_AES_CM_128_HMAC_SHA1_80:
>                 p->cipher_type = AES_128_ICM;
>                 p->cipher_key_len = 30;
>                 p->auth_type = HMAC_SHA1;
>                 p->auth_key_len = 20;
>                 p->auth_tag_len = 10;
>                 p->sec_serv = sec_serv_conf_and_auth;
>                 return 0;
>
>         case AST_AES_CM_128_HMAC_SHA1_32:
>                 p->cipher_type = AES_128_ICM;
>                 p->cipher_key_len = 30;
>                 p->auth_type = HMAC_SHA1;
>                 p->auth_key_len = 20;
>                 p->auth_tag_len = 4;
>                 p->sec_serv = sec_serv_conf_and_auth;
>                 return 0;
>
>         default:
>                 ast_log(LOG_ERROR, "Invalid crypto suite: %u\n", suite);
>
>
>
>
> ------------------------------------------------------------------------------
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> consuming the most bandwidth. Provides multi-vendor support for NetFlow,
> J-Flow, sFlow and other flows. Make informed decisions using capacity
> planning reports. https://ad.doubleclick.net/ddm/clk/305295220;132659582;e
> _______________________________________________
> Srtp-users mailing list
> Srt...@li...
> https://lists.sourceforge.net/lists/listinfo/srtp-users
> .
>

[Srtp-users] Stronger ciphers for Asterisk

[Kevin L. <kev...@ha...>]

Hi folks! First post to the list.


I would like to get more (stronger) cipher suites for SRTP allowed in Asterisk.  I am not really C programmer however, and have never submitted a patch to Asterisk.

 It seems that many modern SIP endpoints are supporting more ciphers. In my case I am using a SIP app called Groundwire, which can use
all the following:

AES_CM_128_HMAC_SHA1_32
AES_CM_128_HMAC_SHA1_80
AES_CM_192_HMAC_SHA1_32
AES_CM_192_HMAC_SHA1_80
AES_CM_256_HMAC_SHA1_32
AES_CM_256_HMAC_SHA1_80
AEAD_AES_128_GCM
AEAD_AES_256_GCM


Asterisk apparently allows only 2  cipher suites for SRTP , both 128 bit .

Below is a snippet from the latest Asterisk stable release tarball - I believe this code section has not been updated in quite some time.   My question is .. could a patch to Asterisk potentially be as simple as just changing this switch/case statement for more options? Or does libSRTP even support the stronger cipher suites listed above?









static int policy_set_suite(crypto_policy_t *p, enum ast_srtp_suite suite)
{
       switch (suite) {
       case AST_AES_CM_128_HMAC_SHA1_80:
               p->cipher_type = AES_128_ICM;
               p->cipher_key_len = 30;
               p->auth_type = HMAC_SHA1;
               p->auth_key_len = 20;
               p->auth_tag_len = 10;
               p->sec_serv = sec_serv_conf_and_auth;
               return 0;

       case AST_AES_CM_128_HMAC_SHA1_32:
               p->cipher_type = AES_128_ICM;
               p->cipher_key_len = 30;
               p->auth_type = HMAC_SHA1;
               p->auth_key_len = 20;
               p->auth_tag_len = 4;
               p->sec_serv = sec_serv_conf_and_auth;
               return 0;

       default:
               ast_log(LOG_ERROR, "Invalid crypto suite: %u\n", suite);

[Srtp-users] SRTP Authentication failed in packet with payload of odd bytes

[Saumil M. <sau...@ma...>]

On Wed, Dec 30, 2015 at 11:38 AM, Saumil Mehta <
sau...@ma...> wrote:

> Hello all,
> I am using mcu opensource conference server provided by medooze to create
> conference.
> That conference module uses libsrtp library with version 1.4.4
>
> I am using Grandstream GXV3140 and Grandstream GXV3175 phones for video
> conference.
> SRTP video call with Grandstream GXV3175 phone is successful and video
> quality is acceptable
> but with Grandstream GXV3140, video is improper and in the logs of mcu
> server it is showing error showing *srtp_unprotect() library function
> returns with error code 7(srtp_error_auth_fail)* in some(not all) packets.
>
> I have checked peer to peer SRTP video call using this both phones and
> there is no issue seen.
>
> I have verified Key is exchanged properly and RTP packet is also given
> properly to library function srtp_unprotect(). Algotrithm used is
> AES_CM_128_HMAC_SHA1_80
>
> *Source of error : Authentication tag came in packet(auth_tag) and
> authentication tag generated by libsrtp(tmp_tag) are mismatched with 1 byte
> shifted*
>
> *i.e. Authentication tag came in packet = 1A 2B 3C 4D 5E and
> authentication tag generated by libsrtp = 2B 3C 4D 5E 6F *(Verified by
> printing them)
>
> *Also noted that this error comes only in the packets having odd number of
> bytes in paylaod.*
>
> PCAP file of same is attached with this mail if needed.
>
> Details : 192.168.101.14 -> mcu conference server IP
>
>               192.168.1.167 -> Grandtstream GXV3140 phone IP
>
> Kindly help me to resolve this issue.
>
> Thank you in advance for your help.
>

Re: [Srtp-users] is it possible to modify srtp headers after encryption?

[Cullen J. (fluffy) <fl...@ci...>]

SRTP does integrity over the whole RTP packet (both headers and payload) but only encrypts the payload. So you can not change any of the headers after encryption. You may be interested in the PERC work at IETF that would allow things like SSRC to change in some cases. 


> On Nov 11, 2015, at 3:11 PM, Mehdi Semsarzadeh <meh...@ma...> wrote:
> 
> Hi there,
> I have a working version of libsrtp which protects and un-protects packets at sender and receiver sides. In the encryption process I send the whole packet to the srtp_protect() method.
> 
> My question is about the encryption algorithm and whether it uses packet header for encrypting the payload data or not?. In other words, if I modify some packet headers (including ssrc) after encrypting process, I get Err: 7 at the receiver end, which is authentication failure! 
> 
> Thanks in advances,
> Mehdi
> 
> ------------------------------------------------------------------------------
> _______________________________________________
> Srtp-users mailing list
> Srt...@li...
> https://lists.sourceforge.net/lists/listinfo/srtp-users

Re: [Srtp-users] is it possible to modify srtp headers after encryption?

[Justin U. <ju...@go...>]

You can't modify the packet after you protect() it, or else the MAC won't
verify.

On Thu, Nov 12, 2015 at 7:11 AM, Mehdi Semsarzadeh <
meh...@ma...> wrote:

> Hi there,
> I have a working version of libsrtp which protects and un-protects packets
> at sender and receiver sides. In the encryption process I send the whole
> packet to the srtp_protect() method.
>
> My question is about the encryption algorithm and whether it uses packet
> header for encrypting the payload data or not?. In other words, if I modify
> some packet headers (including ssrc) after encrypting process, I get Err: 7
> at the receiver end, which is authentication failure!
>
> Thanks in advances,
> Mehdi
>
>
>
> ------------------------------------------------------------------------------
>
> _______________________________________________
> Srtp-users mailing list
> Srt...@li...
> https://lists.sourceforge.net/lists/listinfo/srtp-users
>
>

[Srtp-users] is it possible to modify srtp headers after encryption?

[Mehdi S. <meh...@ma...>]

Hi there,
I have a working version of libsrtp which protects and un-protects packets
at sender and receiver sides. In the encryption process I send the whole
packet to the srtp_protect() method.

My question is about the encryption algorithm and whether it uses packet
header for encrypting the payload data or not?. In other words, if I modify
some packet headers (including ssrc) after encrypting process, I get Err: 7
at the receiver end, which is authentication failure!

Thanks in advances,
Mehdi

Re: [Srtp-users] jitter statistics from libsrtp

[John F. <fo...@ci...>]

Jitter statistics would come from the RTP layer.  For instance, the DSP
processing the RTP packets would generate the statistics. 


On 09/10/2015 08:33 AM, Priyaranjan Nayak wrote:
> Hi All,
>
> I am using libsrtp-1.5.2 version source code for DTLS-SRTP in my
> project and it is working fine without any issue.Is there any way to
> get the jitter statistics of rtp packets from libsrtp ?
>
>
> Thanks
> Priyaranjan

[Srtp-users] jitter statistics from libsrtp

[Priyaranjan N. <pri...@gm...>]

Hi All,

I am using libsrtp-1.5.2 version source code for DTLS-SRTP in my project
and it is working fine without any issue.Is there any way to get the jitter
statistics of rtp packets from libsrtp ?


Thanks
Priyaranjan

[Srtp-users] Audio quality is not good after 10sec of call

[Priyaranjan N. <pri...@gm...>]

Hi David,

I am using libsrtp-1.5.2 vesion. The ouput audio quality is very bad after
10 second of call duration. I am using crypto-suite AES_CM_128_HMAC_SHA1_80
and there is no packet loss in the network.

Could you please guide me why the audio quality is going bad after 10
second of call ?

Thanks
Priyaranjan

[Srtp-users] un-protect() API of libsrtp-1.5.2 is corrupting some packets

[Priyaranjan N. <pri...@gm...>]

Hi All,

I am using libsrtp-1.5.2 version in our project. I have set the
encrypt/decrypt key properly. I am observing below point

   - The rtp packets are same after protect and before unprotect .But when
   I call un-protect() , 80% packets are unprotect properly. The rest of 20%
   packets are not unprotect properly. In these packets some of the bytes gets
   corrupted during un-protect() API call.

If anyone faced this problem in un-protect() API, Please revert back, what
needs to be done to get correct packets.

Thanks
Priyaranjan

[Srtp-users] DTLS-SRTP Key exchange between two SIP endpoint

[Priyaranjan N. <pri...@gm...>]

Hi All,

I am new to the DTLS-SRTP environment. I have two sip endpoint and wanted
to establish the dtls-srtp for media(RTP) communication.
Could you please explain me How the master key and salt key will be
exchange between two sip endpoint while establishing the media in DTLS-SRTP
?


Thanks
Priyaranjan

[Srtp-users] Need some information regarding dtls-srtp

[Priyaranjan N. <pri...@gm...>]

Hi All,

I have only dtls and srtp supported code. I wanted to use both module for
dtls-srtp.
1. Is it possible to combine both module for dtls-srtp ?
2. if possible , Could you please give me idea how can I approach ?

Thanks
Priyaranjan

Re: [Srtp-users] Difference between DTLS-SRTP and SRTP packet send over DTLS connections

[Justin U. <ju...@go...>]

DTLS-SRTP uses normal SRTP packets on the wire, with keys extracted from
DTLS. There is no additional overhead.
SRTP packets sent over DTLS have DTLS framing, which adds somewhat
significant overhead.

On Wed, Jul 15, 2015 at 9:14 AM, John Foley <fo...@ci...> wrote:

>  My knowledge of DTLS-SRTP is fairly limited.  But my understanding is the
> signaling protocol (e.g. SIP) would use DTLS as the transport.  The SRTP
> session parameters are setup via the signaling protocol.  Once those are
> established, SRTP packets are sent directly over UDP (not using DTLS).  But
> you'll probably want to read RFC 5763 to understand the details.
>
>
>
>
> On 07/15/2015 01:14 AM, Priyaranjan Nayak wrote:
>
> I have two SIP endpoint and wanted to send the media over dtls-srtp. I am
> little bit confuse in below points
>
> 1. Is DTLS-SRTP and SRTP packet send over DTLS connections are same or
> different ?
>
> 2. If different, Could you please explain me how are they different ?
>
> Thanks
> Priyaranjan
>
>
>
>
> ------------------------------------------------------------------------------
> Don't Limit Your Business. Reach for the Cloud.
> GigeNET's Cloud Solutions provide you with the tools and support that
> you need to offload your IT needs and focus on growing your business.
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> _______________________________________________
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> Srt...@li...
> https://lists.sourceforge.net/lists/listinfo/srtp-users
>
>

Re: [Srtp-users] Difference between DTLS-SRTP and SRTP packet send over DTLS connections

[John F. <fo...@ci...>]

My knowledge of DTLS-SRTP is fairly limited.  But my understanding is
the signaling protocol (e.g. SIP) would use DTLS as the transport.  The
SRTP session parameters are setup via the signaling protocol.  Once
those are established, SRTP packets are sent directly over UDP (not
using DTLS).  But you'll probably want to read RFC 5763 to understand
the details.



On 07/15/2015 01:14 AM, Priyaranjan Nayak wrote:
> I have two SIP endpoint and wanted to send the media over dtls-srtp. I
> am little bit confuse in below points
>
> 1. Is DTLS-SRTP and SRTP packet send over DTLS connections are same or
> different ?
>
> 2. If different, Could you please explain me how are they different ?
>
> Thanks
> Priyaranjan

[Srtp-users] Difference between DTLS-SRTP and SRTP packet send over DTLS connections

[Priyaranjan N. <pri...@gm...>]

I have two SIP endpoint and wanted to send the media over dtls-srtp. I am
little bit confuse in below points

1. Is DTLS-SRTP and SRTP packet send over DTLS connections are same or
different ?

2. If different, Could you please explain me how are they different ?

Thanks
Priyaranjan

Re: [Srtp-users] How to generate libsrtp.so

[John F. <fo...@ci...>]

Try "make shared_library"


On 07/11/2015 03:30 AM, Priyaranjan Nayak wrote:
> Hi John,
>
> I have tried your command,but I am getting below output for "make
> libsrtp2.so" command
> make: *** No rule to make target `libsrtp2.so'.  Stop.
> I have gone through the Makefile and put command "make
> shared_library". Now I am getting libsrtp.so.1
>
> Is it correct way of generating the shared library for libsrtp-1.5.2 ?
>
> On Fri, Jul 10, 2015 at 7:31 PM, John Foley <fo...@ci...
> <mailto:fo...@ci...>> wrote:
>
>     You should be able to simply do:
>
>     ./configure
>     make clean
>     make libsrtp2.so
>
>
>
>
>     On 07/10/2015 09:50 AM, Priyaranjan Nayak wrote:
>>     Hi All,
>>
>>     I am using libsrtp-1.5.2 version. I wanted to use as a shared
>>     library. I did not get any configuration argument to generate the
>>     shared library.It is generating static library bydefault. Could
>>     you please help me how to generate shared library(libsrtp.so) ?
>>
>>     Thanks
>>     Priyaranjan
>>
>>
>>     ------------------------------------------------------------------------------
>>     Don't Limit Your Business. Reach for the Cloud.
>>     GigeNET's Cloud Solutions provide you with the tools and support that
>>     you need to offload your IT needs and focus on growing your business.
>>     Configured For All Businesses. Start Your Cloud Today.
>>     https://www.gigenetcloud.com/
>>
>>
>>     _______________________________________________
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>>     Srt...@li... <mailto:Srt...@li...>
>>     https://lists.sourceforge.net/lists/listinfo/srtp-users
>
>
>
>
> -- 
> Thanks
> Priyaranjan