[OpenSIPS-Users] Change in SDP/RTP routing with Opensips 1.4 and 1.6

[Chris Stone]

Ovidiu,

On Mon, Feb 7, 2011 at 11:22 AM, Ovidiu Sas <osas at voipembedded.com> wrote:
> Just capture a call that is going through your SIP proxy and check the
> SDP in the received and sent INVITE and the SDP in the received and
> sent 200ok.  The connection IP and port for each SDP should be the
> same (untouched).
> If it's untouched, then your opensips config is working as expected
> and your RTP problem is somewhere else.

As I was seeing traffic wise, OpenSIPS is modifying the INVITE and
putting it's own IP where my 1.4 installation is putting my upstream
provider's IP. Both of these are the SIP body text showing what I am
referring to. Both are inbound calls. First, to our OpenSIPS 1.4
installation from upstream and from that installation to our Asterisk
box (this is working as we want):

    v=0\r\n
    o=PVG 1297116052450 1297116052450 IN IP4 199.173.80.118\r\n
    s=-\r\n
    p=+1 6135555555\r\n
    c=IN IP4 199.173.80.118\r\n
    t=0 0\r\n
    m=audio 52056 RTP/AVP 18 0 8 101\r\n
    a=rtpmap:101 telephone-event/8000\r\n
    a=fmtp:101 0-15\r\n
    a=ptime:20\r\n
    a=fmtp:18 annexb=no\r\n


    v=0\r\n
    o=PVG 1297116052450 1297116052450 IN IP4 199.173.80.118\r\n
    s=-\r\n
    p=+1 6135555555\r\n
    c=IN IP4 199.173.80.118\r\n
    t=0 0\r\n
    m=audio 52056 RTP/AVP 18 0 8 101\r\n
    a=rtpmap:101 telephone-event/8000\r\n
    a=fmtp:101 0-15\r\n
    a=ptime:20\r\n
    a=fmtp:18 annexb=no\r\n

Note that 199.173.80.118 is my upstream provider.

And then from upstream to our OpenSIPS 1.6 installation and from that
installation to our Asterisk box (not working right - SDP is being
relayed via OpenSIPS):

    v=0\r\n
    o=Acme_UAS 0 1 IN IP4 208.94.157.10\r\n
    s=SIP Media Capabilities\r\n
    c=IN IP4 208.94.157.10\r\n
    t=0 0\r\n
    m=audio 25682 RTP/AVP 0 18 101\r\n
    a=rtpmap:0 PCMU/8000\r\n
    a=rtpmap:18 G729/8000\r\n
    a=rtpmap:101 telephone-event/8000\r\n
    a=maxptime:20\r\n
    a=sendrecv\r\n

    v=0\r\n
    o=Acme_UAS 0 1 IN IP4 67.112.153.182\r\n
    s=SIP Media Capabilities\r\n
    c=IN IP4 67.112.153.182\r\n
    t=0 0\r\n
    m=audio 25682 RTP/AVP 0 18 101\r\n
    a=rtpmap:0 PCMU/8000\r\n
    a=rtpmap:18 G729/8000\r\n
    a=rtpmap:101 telephone-event/8000\r\n
    a=maxptime:20\r\n
    a=sendrecv\r\n

Note that 208.94.157.1 is my upstream provider and 67.112.153.182 is
the OpenSIPS 1.6 server.

So, yes, looks like the issue is in Opensips. As I noted though, we're
using the same config with 1.6 (posted in previous message) as we have
running on 1.4. So not sure why now OpenSIPS is modifying and putting
itself in the relay path for SDP when we only want that for SIP as
with 1.4.

Thanks and any ideas? Is there some default for this that may have
changed from 1.4 to 1.6 that I missed in the upgrade docs?


Best Regards,

Chris