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<font size="-1"><font face="Arial">hi,<br>
<br>
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.<br>
<br>
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.<br>
<br>
any idea what could be wrong<br>
<br>
my dialplan looks like this<br>
<br>
<extension name="unloop"><br>
<condition field="${unroll_loops}"
expression="^true$"/><br>
<condition field="${sip_looped_call}"
expression="^true$"><br>
<action application="deflect"
data="${destination_number}"/><br>
</condition><br>
</extension><br>
<br>
<extension name="general_stuff" continue="true"><br>
<condition><br>
<action application="set"
data="suppress_cng=true"/><br>
</condition><br>
</extension><br>
<br>
<extension name="local_call"><br>
<condition field="destination_number"
expression="^000$"><br>
<action application="export"
data="rtp_secure_media=true"/><br>
<action application="bridge"
data="${sofia_contact(${vps_id})}"/><br>
</condition><br>
</extension><br>
<br>
my porfile looks like this<br>
<br>
<param name="inbound-bypass-media" value="false"/><br>
<param name="inbound-late-negotiation" value="false"/><br>
<param name="inbound-zrtp-passthru" value="false"/><br>
<param name="disable-transcoding" value="true"/><br>
<param name="vad" value="none"/><br>
<param name="rtp-autofix-timing" value="false"/><br>
<param name="rtp-rewrite-timestamps" value="false"/><br>
<param name="send-display-update" value="false"/><br>
<param name="accept-blind-reg" value="false"/><br>
<br>
<br>
</font></font>
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</style>Regards,<br>
Bipin<br>
<br>
<br>
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