[Users] changing SDP session and media attributes in 200 OK - sr-users

24 Apr 2006


      Hello,
i am using rtpproxy with openser to act as a RTP bridge to IP-SS7 gateway. IP-SS7 GW should not be exposed to external clients directly. 
Architecture diagram is like this:
|UA1|--SIP/RTP---|
          |------------------|             |-----------|  
          | OpenSer+RTPProxy | --SIP/RTP-- | IP-SS7-GW | 
          |------------------|             |-----------|
|UA2|--SIP/RTP---|
Problem is when i try to send fax RTP packets destined for UA2 from IP-SS7 GW are sent directly. 
They should go through RTPProxy, otherwise media ports does not  match.
onreply route looks like:
onreply_route[1] {
    if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") {
         fix_nated_contact();
    }
    force_rtp_proxy();
}
each request is record_route()'ed before it is sent to be routed back through proxy.
I noticed that OpenSer removes top most via header in 200 OK  messages:
200 OK message for re-Invite received by OpenSer:
SIP/2.0 200 OK\r\n
    From: sip:870000049@sp2.ntt.lt;tag=4fbb0552\r\n
    To: sip:70000050@sp2.ntt.lt;tag=0-13c4-444c9fc6-f4eac-fdd\r\n
    Call-ID: cc36b0-0-13c4-444c9fc6-f4eac-2c87@sp2.ntt.lt\r\n
    CSeq: 666497904 INVITE\r\n
    Via: SIP/2.0/UDP 84.32.0.18;branch=z9hG4bKf67f.2aeace41.0\r\n
    Via: SIP/2.0/UDP 84.32.0.7:5060;branch=z9hG4bK444cbbfca6214604dc9\r\n
    Supported: 100rel\r\n
    User-Agent: SIP-RG\r\n
    Contact: sip:70000050@84.32.134.143\r\n
    Content-Type: application/sdp\r\n
    Content-Length: 317\r\n
    \r\n
    v=0\r\n
    o=SIP-N-TA 0 0 IN IP4 84.32.134.143\r\n
    s=5420868F 0000 5420868F 0000 01 0\r\n
    c=IN IP4 84.32.134.143\r\n
    t=0 0\r\n
    m=image 4008 udptl t38\r\n
    c=IN IP4 84.32.134.143\r\n
    a=T38FaxVersion:0\r\n
    a=T38MaxBitRate:14400\r\n
    a=T38FaxFillBitRemoval:0\r\n
    a=T38FaxTranscodingMMR:0\r\n
    a=T38FaxTranscodingJBIG:0\r\n
    a=T38FaxRateManagement:transferredTCF\r\n
200 OK message sent to IP-SS7 GW by OpenSer:
SIP/2.0 200 OK
From: sip:870000049@sp2.ntt.lt;tag=4fbb0552
To: sip:70000050@sp2.ntt.lt;tag=0-13c4-444c9fc6-f4eac-fdd
Call-ID: cc36b0-0-13c4-444c9fc6-f4eac-2c87@sp2.ntt.lt
CSeq: 666497904 INVITE
Via: SIP/2.0/UDP 84.32.0.7:5060;branch=z9hG4bK444cbbfca6214604dc9
Supported: 100rel
User-Agent: SIP-RG
Contact: sip:70000050@84.32.134.143
Content-Type: application/sdp
Content-Length: 333
v=0
o=SIP-N-TA 0 0 IN IP4 84.32.134.143
s=5420868F 0000 5420868F 0000 01 0
c=IN IP4 84.32.134.143
t=0 0
m=image 39246 udptl t38
c=IN IP4 84.32.0.18
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=nortpproxy:yes
In this message there is no indication that is came from SIP proxy (no record-route, no via header), then IP-SS7 GW sends ACK message directly.
This, i suppose, is normal thing, is it? Problem is that IP-SS7 also sends RTP traffic to UA2 directly. This probably happens because of two 'c=' fields in SDP.
OpenSer changes only one in  media part. Is there a way to change 'c=' field in SDP not only on media part? 
Which should take precedence if both can't be changed?
regards,
Antanas Masevicius