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<div class="moz-cite-prefix"> Does your client see each other can
UDP traffic pass from one client to the other? You mentioned any
connect so my guess is you have a cisco router/firewall before FS
and all your problems reside there. We have very similar setup
that actually works with bypass media true.<br>
<br>
On 9/10/2013 5:15 AM, Peter wrote:<br>
</div>
<blockquote
cite="mid:CABkS=xrje-Vye0opXcxqcfjJNiewJUXW9yoo3j75PUDW2VakgQ@mail.gmail.com"
type="cite">
<div dir="ltr">
<div>My goal is to have SIP and media go over the split tunnel
VPN, with the SIP traffic going through FS (which it does) and
the RTP stream between the VPN IP addresses of the two
endpoints.<br>
<br>
</div>
I have inbound-bypass-media set to true as I don't want the
media to go via FS.<br>
<br>
Does that make sense?<br>
</div>
<div class="gmail_extra"><br>
<br>
<div class="gmail_quote">On Mon, Sep 9, 2013 at 10:09 PM,
Michael Jerris <span dir="ltr"><<a moz-do-not-send="true"
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">
<div style="word-wrap:break-word">are you trying to send the
media over the vpn now or peer to peer? do you still
have inbound-bypass-media set to true? Peer to peer may
never work depending on what is in the original sdp of the
clients.
<div>
<div class="h5">
<div><br>
<div>
<div>On Sep 8, 2013, at 11:28 PM, Peter <<a
moz-do-not-send="true"
href="mailto:eidevm5@gmail.com"
target="_blank">eidevm5@gmail.com</a>>
wrote:</div>
<br>
<blockquote type="cite">
<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
moz-do-not-send="true"
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>
<div><<a moz-do-not-send="true"
href="mailto:beskrovny@gmail.com"
target="_blank">beskrovny@gmail.com</a>>
wrote:<br>
<br>
> <a moz-do-not-send="true"
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
moz-do-not-send="true"
href="mailto:eidevm5@gmail.com"
target="_blank">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
moz-do-not-send="true"
href="mailto:sips%3A1004@10.1.1.204"
target="_blank">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
moz-do-not-send="true"
href="mailto:sip%3A1001@10.1.1.204"
target="_blank">sip:1001@10.1.1.204</a>>;tag=cN83ZcmHw<br>
>> To: <a
moz-do-not-send="true"
href="mailto:sip%3A1004@10.1.1.204"
target="_blank">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
moz-do-not-send="true"
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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<pre wrap="">_________________________________________________________________________
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