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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