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 <http://example.com> gateway sip:joeuser@example.com <sip%3Ajoeuser@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 <http://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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-- *Thanks:* Saurabh Kumar Verma VVDN Technologies Pvt Ltd *Cell* : +91 7042378747 | *Skype* : saurabh.verma001