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<font size="-1"><font face="Arial">hi,<br>
<br>
dont bother, i managed to figure this out, what i needed to do
was export absolute_codec_string with PCMU before the bridge<br>
<br>
<br>
</font></font>
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</style>Regards,<br>
Bipin<br>
<br>
<br>
<hr>
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<div class="moz-cite-prefix">-------- Original Message --------<br>
Subject: [Freeswitch-users] tls srtp calls between users white
noise<br>
From: Bipin Patel <a class="moz-txt-link-rfc2396E" href="mailto:bipin@xbipin.com"><bipin@xbipin.com></a><br>
To: FreeSWITCH Users Help
<a class="moz-txt-link-rfc2396E" href="mailto:freeswitch-users@lists.freeswitch.org"><freeswitch-users@lists.freeswitch.org></a><br>
Date: 9/23/2017, 1:26:52 AM<br>
</div>
<blockquote type="cite"
cite="mid:8a5988df-e823-05fb-562e-5bb0620d068d@xbipin.com">
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charset=windows-1252">
<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>
<div class="moz-signature">-- <br>
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p.footr { font-family: "Trebuchet MS", Verdana, sans-serif; margin:0in; margin-bottom:.0001pt; }
h1 { font-size: 1.2em; }
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a, a:link, a:visited { color: #2A5685;}
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a.wiki-anchor { display: none; }
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}
</style>Regards,<br>
Bipin<br>
<br>
<br>
<hr> </div>
<br>
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<br>
<pre wrap="">_________________________________________________________________________
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