[Users] Error in Radius Start when attemping write SDP
Bogdan-Andrei Iancu
bogdan at voice-system.ro
Wed Mar 15 10:32:33 CET 2006
Hi Dmitry,
that's very interesting what you are reporting. As there is no
difference if the call comes from a SIP phone or a GW from request
processing point of view, I would say the cause may be the request
itself - do you get a prior error messages for the request from GW?
regards,
bogdan
Dmitry Lyubimkov wrote:
>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 at 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 at voapp.ru:5060>;tag=0013464F617E_T195292895
>To: <sip:78142799111 at voapp.ru:5060>
>Call-ID: CALL_ID17_0013464F617E_T1817335404 at 172.16.224.5
>CSeq: 2100902695 INVITE
>Contact: <sip:loftphone at 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 at voapp.ru"
> Calling-Station-Id = "sip:loftphone at voapp.ru:5060"
> Called-Station-Id = "sip:78142799111 at voapp.ru:5060"
> Sip-Translated-Request-URI = "sip:78142799111 at voapp.ru:5060"
> Acct-Session-Id =
>"CALL_ID17_0013464F617E_T1817335404 at 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 at voapp.ru:5060 SIP/2.0
>Via: SIP/2.0/UDP 62.33.22.11:5060;x-route-tag="tgrp:ipphone"
>From: <sip:78142760321 at 62.33.22.11>;tag=97134E7C-1EF1
>To: <sip:78142799299 at voapp.ru>
>Date: Mon, 13 Mar 2006 20:12:09 GMT
>Call-ID: 7BF31E00-B20411DA-85D59372-594AF1F at 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 at 62.33.22.11>;party=calling;screen=yes;privacy=
>off
>Timestamp: 1142280729
>Contact: <sip:78142760321 at 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
>
>
>
>
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