<div dir="ltr"><div>Hi Michael.<br><br></div>I have rtp-ip set to the VPN IP address. Is there any other setting I can try?<br><br>Thanks<br><br>Peter<br></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Fri, Sep 6, 2013 at 4:24 PM, Michael Jerris <span dir="ltr"><<a href="mailto:mike@jerris.com" target="_blank">mike@jerris.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">if your talking about the media still hitting fs but through a<br>
different ip that isn't going through VPN, you should be able to work<br>
this out by adjusting rtp-ip param<br>
<br>
On Sep 5, 2013, at 7:52 AM, "Veniamin (Benjamin) Beskrovny"<br>
<div class="HOEnZb"><div class="h5"><<a href="mailto:beskrovny@gmail.com">beskrovny@gmail.com</a>> wrote:<br>
<br>
> <a href="http://wiki.freeswitch.org/wiki/Codec_negotiation#Rewriting_SDP" target="_blank">http://wiki.freeswitch.org/wiki/Codec_negotiation#Rewriting_SDP</a> maybe?<br>
><br>
> On 9/5/13, Peter <<a href="mailto:eidevm5@gmail.com">eidevm5@gmail.com</a>> wrote:<br>
>> This is not a direct FreeSWITCH question, but was hoping someone might have<br>
>> set up a similar configuration.<br>
>><br>
>> Currently I have a FS server where SIP clients (Linphone) register to via a<br>
>> Cisco AnyConnect VPN. I have inbound-bypass-media set to true so that RTP<br>
>> traffic is peer to peer.<br>
>><br>
>><br>
>> If I tunnel all traffic, the SIP transport and RTP media works just fine.<br>
>><br>
>> If I set up a VPN split tunnel that allows the subnets of the VPN client<br>
>> and the FS server, the SIP signalling is fine, but in the SDP, it specifies<br>
>> a non VPN address, eg:<br>
>><br>
>> INVITE <a href="mailto:sips%3A1004@10.1.1.204">sips:1004@10.1.1.204</a>;transport=udp SIP/2.0<br>
>> Via: SIP/2.0 10.1.254.36:53036;branch=z9hG4bK.GRrzYXeCN;rport<br>
>> From: <<a href="mailto:sip%3A1001@10.1.1.204">sip:1001@10.1.1.204</a>>;tag=cN83ZcmHw<br>
>> To: <a href="mailto:sip%3A1004@10.1.1.204">sip:1004@10.1.1.204</a><br>
>> CSeq: 20 INVITE<br>
>> Call-ID: e7eEYxoML2<br>
>> Max-Forwards: 70<br>
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,<br>
>> SUBSCRIBE, INFO<br>
>> Content-Type: application/sdp<br>
>> Content-Length: 426<br>
>> User-Agent: LinphoneAndroid/2.1.2-177-g015b83c (belle-sip/0.0.1)<br>
>> Contact: <<a href="http://sip:1001@10.1.254.36:53036" target="_blank">sip:1001@10.1.254.36:53036</a><br>
>> ;transport=udp>;+sip.instance="<urn:uuid:83c4b823-9c0d-4e02-b7f7-cdd3ae2f1259>"<br>
>><br>
>> v=0<br>
>> o=1001 3921 1865 IN IP4 [ISP address]<br>
>> s=Talk<br>
>> c=IN IP4 [ISP address]<br>
>> b=AS:380<br>
>> t=0 0<br>
>> m=audio 1031 RTP/SAVP 0 8 100 101<br>
>> a=rtpmap:100 iLBC/8000<br>
>> a=fmtp:100 mode=30<br>
>> a=rtpmap:101 telephone-event/8000<br>
>> a=fmtp:101 0-11<br>
>> a=rtcp:7079 IN IP4 [wireless IP address]<br>
>><br>
>><br>
>> I'm not very familiar with how the SDP is constructed, but I'm assuming<br>
>> that it's the SIP client that's responsible for offering it's IP addresses.<br>
>><br>
>> Are there any FreeSWITCH settings that would affect the IP addresses in the<br>
>> SDP?<br>
>><br>
>> Thanks<br>
>><br>
>> Peter<br>
><br>
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</div></div></blockquote></div><br></div>