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.I have inbound-bypass-media set to true as I don't want the media to go via FS.
Does that make sense?_________________________________________________________________________On Mon, Sep 9, 2013 at 10:09 PM, Michael Jerris <mike@jerris.com> wrote:
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@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@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@gmail.com> wrote:
> http://wiki.freeswitch.org/wiki/Codec_negotiation#Rewriting_SDP maybe?
>
> On 9/5/13, Peter <eidevm5@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@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@10.1.1.204>;tag=cN83ZcmHw
>> To: sip:1004@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@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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