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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