[Freeswitch-users] RTP Over VPN Split Tunnel

[Michael Jerris]

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.

On Sep 8, 2013, at 11:28 PM, Peter <eidevm5 at gmail.com> wrote:

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