Hi,

So audio is working properly until we don't set these following parameters (Changes has been done because we don't able to see secure parameter is set in channels table & it's set only for caller's endpoint).

i). Change in default.xml in dialplan:
<condition field="${rtp_has_crypto}" expression="^($${rtp_sdes_suites})$" break="never">
        <action application="set" data="rtp_secure_media=true"/>
        <!-- Offer SRTP on outbound legs if we have it on inbound. -->
        <action application="export" data="rtp_secure_media=true"/>
      </condition>

      <!--
         Since we have inbound-late-negotation on by default now the
         above behavior isn't the same so you have to do one extra step.
        -->
      <condition field="${endpoint_disposition}" expression="^(DELAYED NEGOTIATION)"/>
      <condition field="${switch_r_sdp}" expression="(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)" break="never">
        <action application="set" data="rtp_secure_media=true"/>
        <!-- Offer SRTP on outbound legs if we have it on inbound. -->
        <action application="export" data="rtp_secure_media=true"/>
      </condition>


ii). Set inbound-late-megotiation to false in sip_profiles/internal.xml..

<param name="inbound-late-negotiation" value="false"/>

On Fri, Sep 11, 2015 at 7:30 AM, Giovanni Maruzzelli <gmaruzz@gmail.com> wrote:

I would counseil to start from the dafault config, and change one thing at time.
See at which change it breaks.

sent from my mobile,
Giovanni Maruzzelli
cell: +39 347 266 56 18

On Sep 11, 2015 3:47 AM, "Saurabh Kumar Verma" <saurabhkumar.verma@vvdntech.com> wrote:
Hi, 

Hope you're doing good. 

I'm facing an issue with having FreeSWITCH (TLS + SRTP). 

Issue: Audio not coming when I've SRTP(media) enabled on both of the soft-phone endpoints. 
Observation: Audio is coming if we don't use SRTP for media encryption. 


Set-up Description:
I'm registered using TLS and able to make call through FreeSWITCH, but call is disconnected after 30 sec because there is not media flow between the endpoints.
I have two softphone (we're using Linphone as soft-phone) registered on FS server with extension 1003 & 1010, call flow is like:

Call Flow
1003 ------> FS Server -------> 1010

x.x.x.x - Server Public IP
y.y.y.y - Server Local IP
NOTE: For security reasons replaced the actual IPs.

FreeSWITCH Version Information:
freeswitch@internal> version
FreeSWITCH Version 1.4.21+git~20150901T202622Z~a223dd0236~64bit (git a223dd0 2015-09-01 20:26:22Z 64bit)

I've done some changes in default dialplan/configuration in FreeSWITCH:

1. Force FreeSWICTH to listen on TLS port only (By using this parameter <param name="tls-only" value="true"/> in internal as well as external profile).

freeswitch@internal> sofia status
                     Name   Type                                      Data State
=================================================================================================
                 external profile           sip:mod_sofia@x.x.x.x:5081 RUNNING (0) (TLS)
    external::example.com gateway                   sip:joeuser@example.com NOREG
             x.x.x.x  alias                                  internal ALIASED
                 internal profile           sip:mod_sofia@x.x.x.x:5061 RUNNING (0) (TLS)
=================================================================================================
2 profiles 1 alias

2. Use the ODBC driver for driver (By uncomment this <!--<param name="odbc-dsn" value="dsn:user:pass"/>-->  in config files)


In Linphone soft-client, these are the changes has been done:
1. TLS as Transport.
2. SRTP in Media encryption.


In FreeSWITCH, these are the registration logs showing I'm softphones are using TLS
freeswitch@internal> sofia status profile internal reg

Registrations:
=================================================================================================
Call-ID:     xmuVdotDkb
User:       1003@x.x.x.x
Contact:     "" <sip:1003@180.151.83.178:63093;app-id=622464153529;pn-type=google;pn-tok=APA91bF-MsdZGvDi951jjCoTDSJc2reyR2JvYHlgtwpeE1vLAlG2zrOIrBmwzx6PPPAHKfcE8aqaOHFtYFoLVztBeqAqz9Cr6D6waN7VCWSIQk7dAdm9HXQ;transport=tls>
Agent:       LinphoneAndroid/2.4.1-28-g98516d9 (belle-sip/1.4.1)
Status:     Registered(TLS)(unknown) EXP(2015-09-09 11:16:30) EXPSECS(3137)
Ping-Status: Reachable
Host:       ip-y-y-y-y.ec2.internal
IP:         180.151.83.178
Port:       63093
Auth-User:   1003
Auth-Realm: x.x.x.x
MWI-Account: 1003@x.x.x.x

Call-ID:     6B25YNRXb5
User:       1010@x.x.x.x
Contact:     "" <sip:1010@180.151.83.178:13916;app-id=622464153529;pn-type=google;pn-tok=APA91bFqivAK_KIDpU_6PM0pf0U8rx9DOKm0vhyNRqjE1Dpq_uPRbTbT-BMwxNP5NmEyCMfnKxa-fjEhI2J-lzLkCcfFphO1hL39cE4VNqAnnfDbVeQbvmQ;transport=tls>
Agent:       LinphoneAndroid/2.4.1-28-g98516d9 (belle-sip/1.4.1)
Status:     Registered(TLS)(unknown) EXP(2015-09-09 11:11:31) EXPSECS(2838)
Ping-Status: Reachable
Host:       ip-y-y-y-y.ec2.internal
IP:         180.151.83.178
Port:       13916
Auth-User:   1010
Auth-Realm: x.x.x.x
MWI-Account: 1010@x.x.x.x

Total items returned: 2
=================================================================================================

For an active call I can see in the channels table, I'm able to see secure parameter is set during the call for both of the call legs.

freeswitch=# select * from channels ;
                 uuid                 | direction |       created       | created_epoch |                   name                   |       state       |    cid_name    | cid_num |    ip_addr 
    | dest | application |    application_data    | dialplan | context | read_codec | read_rate | read_bit_rate | write_codec | write_rate | write_bit_rate |              secure              
 |           hostname           |    presence_id    | presence_data | callstate |  callee_name  | callee_num | callee_direction |              call_uuid               | sent_callee_name | sen
t_callee_num | initial_cid_name | initial_cid_num | initial_ip_addr | initial_dest | initial_dialplan | initial_context 
--------------------------------------+-----------+---------------------+---------------+------------------------------------------+-------------------+----------------+---------+------------
----+------+-------------+------------------------+----------+---------+------------+-----------+---------------+-------------+------------+----------------+----------------------------------
-+------------------------------+-------------------+---------------+-----------+---------------+------------+------------------+--------------------------------------+------------------+----
-------------+------------------+-----------------+-----------------+--------------+------------------+-----------------
 485d07d6-56da-11e5-ac1a-a53fe62ce2f9 | outbound  | 2015-09-09 10:05:22 |    1441793122 | sofia/internal/1010@180.151.83.178:13916 | CS_EXCHANGE_MEDIA | Extension 1003 | 1003    | 180.151.83.
178 | 1010 |             |                        | XML      | default | opus       | 48000     | 0             | opus        | 48000      | 0              | srtp:sdes:AES_CM_256_HMAC_SHA1_80
 | ip-172-31-42-34.ec2.internal | 1010@x.x.x.x |               | ACTIVE    | Outbound Call | 1010       | SEND             | 47fa445c-56da-11e5-abf5-a53fe62ce2f9 | Extension 1003   | 100
3            | Extension 1003   | 1003            | 180.151.83.178  | 1010         | XML              | default
 47fa445c-56da-11e5-abf5-a53fe62ce2f9 | inbound   | 2015-09-09 10:05:22 |    1441793122 | sofia/internal/1003@x.x.x.x         | CS_EXECUTE        | 1003           | 1003    | 180.151.83.
178 | 1010 | bridge      | user/1010@x.x.x.x | XML      | default | opus       | 48000     | 0             | opus        | 48000      | 0              | srtp:sdes:AES_CM_128_HMAC_SHA1_80
 | ip-172-31-42-34.ec2.internal | 1003@x.x.x.x |               | ACTIVE    | Outbound Call | 1010       | SEND             | 47fa445c-56da-11e5-abf5-a53fe62ce2f9 | Outbound Call    | 101
0            | 1003             | 1003            | 180.151.83.178  | 1010         | XML              | default
(2 rows)

 
Attached herewith the complete FreeSWITCH log. Please let me know if any other information is required.

Any help from your side would be really appreciated. Seeking some guidance & help from FreeSWITCH team.

--

Thanks:

Saurabh Kumar Verma

VVDN Technologies Pvt Ltd

Cell : +91 7042378747 | Skype : saurabh.verma001

_________________________________________________________________________
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http://www.freeswitchsolutions.com

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_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

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--

Thanks:

Saurabh Kumar Verma

VVDN Technologies Pvt Ltd

Cell : +91 7042378747 | Skype : saurabh.verma001