We use OpenSER 1.0.1 There is this command in config file: modparam("acc", "radius_extra", "Sip-UA=$hdr(User-Agent); Sip-Via=$hdr(Via[*]); Sip-SDP=$rb")
When the call goes from UA to gateway (Cisco AS5350) we receive this SIP the message (received by tcpdump)
23:37:04.001136 IP 172.16.224.5.5060 > 62.33.22.14.5060: UDP, length 845 E..i....:.......>!.......UU.INVITE sip:78142799111@voapp.ru:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.224.5:5060;branch=z9hG4bK_0013464F617E_T72405C7F Session-Expires: 1800;refresher=uas From: "loftphone" sip:loftphone@voapp.ru:5060;tag=0013464F617E_T195292895 To: sip:78142799111@voapp.ru:5060 Call-ID: CALL_ID17_0013464F617E_T1817335404@172.16.224.5 CSeq: 2100902695 INVITE Contact: sip:loftphone@172.16.224.5:5060 Max-Forwards: 70 Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,DO,UPDATE,OPTIONS,SUBSCRIBE,INFO Supported: timer,replaces User-Agent: DPH-12001.00.04 Content-Type: application/sdp Content-Length: 243
v=0 o=loftphone 416452497 416452497 IN IP4 172.16.224.5 s=DPH-12001.00.04 c=IN IP4 172.16.224.5 t=0 0 m=audio 41000 RTP/AVP 8 18 4 a=rtpmap:8 PCMA/8000/1 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000/1 a =sendrecv
And in Radius we receive following Start record:
Mon Mar 13 23:37:04 2006 Acct-Status-Type = Start Service-Type = Sip-Session Sip-Response-Code = 200 Sip-Method = 1 User-Name = "loftphone@voapp.ru" Calling-Station-Id = "sip:loftphone@voapp.ru:5060" Called-Station-Id = "sip:78142799111@voapp.ru:5060" Sip-Translated-Request-URI = "sip:78142799111@voapp.ru:5060" Acct-Session-Id = "CALL_ID17_0013464F617E_T1817335404@172.16.224.5" Sip-To-Tag = "972A1EA0-2615" Sip-From-Tag = "0013464F617E_T195292895" Sip-Cseq = "2100902696" Sip-UA = "DPH-12001.00.04" Sip-Via = "SIP/2.0/UDP 172.16.224.5:5060;branch=z9hG4bK_0013464F617E_T00 82E2E0" Sip-SDP = "v=0\r\no=loftphone 416452497 416452497 IN IP4 172.16.224.5\r\ ns=DPH-12001.00.04\r\nc=IN IP4 172.16.224.5\r\nt=0 0\r\nm=audio 41000 RTP/AVP 8 18 4\r\na=rtpmap:8 PCMA/8000/1\r\na=rtpmap:18 G729/8000/1\r\na=fmtp:18 annexb=no \r\na=rtpmap:4 G723/8000/1\r\na=sendrecv\r\n" NAS-Port = 5060 Acct-Delay-Time = 0 NAS-IP-Address = 172.16.2.6 Client-IP-Address = 172.16.2.6 Acct-Unique-Session-Id = "20c2ac8753fb6a0d" Stripped-User-Name = "loftphone" Realm = "voapp.ru" Timestamp = 1142282224
But if the call goes from gateway to UA that Start record does not come. Invite message looks as follows:
23:12:08.997977 IP 62.33.22.11.55068 > 62.33.22.14.5060: UDP, length 1119 E..{.......w>!..>!.......g,.INVITE sip:78142799299@voapp.ru:5060 SIP/2.0 Via: SIP/2.0/UDP 62.33.22.11:5060;x-route-tag="tgrp:ipphone" From: sip:78142760321@62.33.22.11;tag=97134E7C-1EF1 To: sip:78142799299@voapp.ru Date: Mon, 13 Mar 2006 20:12:09 GMT Call-ID: 7BF31E00-B20411DA-85D59372-594AF1F@195.161.136.114 Supported: timer,100rel Min-SE: 1800 Cisco-Guid: 2079450544-2986611162-2245170034-93630239 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY , INFO CSeq: 101 INVITE Max-Forwards: 6 Remote-Party-ID: sip:78142760321@62.33.22.11;party=calling;screen=yes;privacy= off Timestamp: 1142280729 Contact: sip:78142760321@62.33.22.11:5060 Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 316
v=0 o=CiscoSystemsSIP-GW-UserAgent 9211 3494 IN IP4 62.33.22.11 s=SIP Call c=IN IP4 62.33.22.11 t=0 0 m=audio 16624 RTP/AVP 3 18 8 0 4 c=IN IP4 62.33.22.11 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:4 G723/8000
a=fmtp:4 annexa=yes
And openser writes in log:
Mar 13 23:41:27 sip OpenSER[18329]: rc_avpair_assign: bad attribute length Mar 13 23:41:27 sip OpenSER[18329]: ERROR: acc_rad_request: rc_avpaid_add failed for (null)
The reason is clear - one attribute is empty. using command modparam("acc", "radius_extra", "Sip-UA=$hdr(User-Agent); Sip-Via=$hdr(Via[*]); Sip-SDP=$rb") we add 3 attribute. Easily we find out that the reason in Sip-SDP = $ rb If to remove this attribute having left only modparam("acc", "radius_extra", "Sip-UA=$hdr(User-Agent); Sip-Via=$hdr(Via[*])") then Start records start to come. But in fact it is perfectly visible that INVITE from a gateway contains SDP. Why pseudo variable $rb does not return value? In what there can be a reason?
Dmitry