tls srtp calls between users white noise

22 Sep 2017

      hi,

i got tls and srtp working between client and FS, client registers to FS 
using TLS and SRTP and if that call is forwarded to gateway in SIP and 
RTP everything works fine.

Problem arises when 2 clients using zoiper register to FS using TLS and 
SRTP and any calls routed between them the calls goes through and the 
phone rings but on answering there is white noise constantly, i checked 
in FS and it does send the call using TLS and SRTP and key exchange and 
codec negotiation all is fine using g711u but i cant seem to get rid of 
this white noise almost making it impossible to each party to hear the 
other.

any idea what could be wrong

my dialplan looks like this

     <extension name="unloop">
       <condition field="${unroll_loops}" expression="^true$"/>
       <condition field="${sip_looped_call}" expression="^true$">
           <action application="deflect" data="${destination_number}"/>
       </condition>
     </extension>

     <extension name="general_stuff" continue="true">
       <condition>
               <action application="set" data="suppress_cng=true"/>
       </condition>
     </extension>

     <extension name="local_call">
       <condition field="destination_number" expression="^000$">
         <action application="export" data="rtp_secure_media=true"/>
         <action application="bridge" data="${sofia_contact(${vps_id})}"/>
       </condition>
     </extension>

my porfile looks like this

     <param name="inbound-bypass-media" value="false"/>
     <param name="inbound-late-negotiation" value="false"/>
     <param name="inbound-zrtp-passthru" value="false"/>
     <param name="disable-transcoding" value="true"/>
     <param name="vad" value="none"/>
     <param name="rtp-autofix-timing" value="false"/>
     <param name="rtp-rewrite-timestamps" value="false"/>
     <param name="send-display-update" value="false"/>
     <param name="accept-blind-reg" value="false"/>

-- 
Regards,
Bipin

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Bipin Patel

Bipin Patel

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