Re: [Opalvoip-user] RE-INVITE is sent without crypto SDP during a SRTP call (and Asterisk)

[Robert J. <ro...@vo...>]

Sorry this took so long, got busy. It should now be fixed.

*Robert Jongbloed*
/OPAL/OpenH323/PTLib Architect and Co-founder./
Commercial support at http://www.voxlucida.com.au

On 5/12/2012 10:53 PM, Diego Busacca wrote:
> Hello,
> I'm using Opal 3.12 (r28646) with Asterisk and TLS/SRTP.
>
> When I try to put a call on hold the RE-INVITE message is sent without
> the crypto SDP line ad so Asterisk reject to put the call on hold
> because the encryption is mandatory.
>
>
> First INVITE (call start):
>
>
> INVITE sip:600@192.168.1.151:5062;transport=tls SIP/2.0
> Route: <sip:192.168.1.151:5062;lr>
> CSeq: 1 INVITE
> Via: SIP/2.0/TLS
> 192.168.0.7:53084;branch=z9hG4bKf0ffa6ac-a508-1910-9705-18037327ae65;rport
> User-Agent: SoftPhone/0.9.2
> From: <sip:2310@192.168.1.151>;tag=fbe4a6ac-a508-1910-9704-18037327ae65
> Call-ID: f5e5a6ac-a508-1910-9704-18037327ae65@dbusacca
> Supported: 100rel,replaces
> To: <sip:600@192.168.1.151>
> Contact: "dbusacca" <sip:2310@192.168.0.7:5063;transport=tls>
> Allow:
> INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING,PRACK
> Content-Length: 276
> Content-Type: application/sdp
> Max-Forwards: 70
>
> v=0
> o=- 1354707170 1 IN IP4 192.168.0.7
> s=TVoxPhone/0.9.2
> c=IN IP4 192.168.0.7
> t=0 0
> m=audio 12000 RTP/SAVP 106
> a=sendrecv
> a=rtpmap:106 ILBC/8000/1
> a=fmtp:106 mode=20
> a=crypto:1 AES_CM_128_HMAC_SHA1_80
> inline:KSO+hOFs1q5SkEnx8bvp67Om2zyHDD6ZJF4NHAa3
> a=maxptime:150
>
>
>
>
> But when I try to put the call on hold:
>
>
> INVITE sip:600@192.168.1.151:5062;transport=TLS SIP/2.0
> Route: <sip:192.168.1.151:5062;lr>
> CSeq: 3 INVITE
> Via: SIP/2.0/TLS
> 192.168.0.7:53084;branch=z9hG4bKc509aaac-a508-1910-9707-18037327ae65;rport
> User-Agent: SoftPhone/0.9.2
> Authorization: Digest username="2310", realm="tvox", nonce="03fa5e39",
> uri="sip:600@192.168.1.151:5062", algorithm=MD5,
> response="50d24d8c0d6f98eb4ef20a4525471ffb"
> From: <sip:2310@192.168.1.151>;tag=fbe4a6ac-a508-1910-9704-18037327ae65
> Call-ID: f5e5a6ac-a508-1910-9704-18037327ae65@dbusacca
> Supported: 100rel,replaces
> To: <sip:600@192.168.1.151>;tag=as62e6f602
> Contact: "dbusacca" <sip:2310@192.168.0.7:5063;transport=tls>
> Allow:
> INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING,PRACK
> Content-Length: 191
> Content-Type: application/sdp
> Max-Forwards: 70
>
> v=0
> o=- 1354707170 2 IN IP4 192.168.0.7
> s=TVoxPhone/0.9.2
> c=IN IP4 192.168.0.7
> t=0 0
> m=audio 12000 RTP/AVP 106
> a=recvonly
> a=rtpmap:106 ILBC/8000/1
> a=fmtp:106 mode=20
> a=maxptime:150
>
>
>
> Then asterisk answer with 488 because SRTP is required.
>
> SIP/2.0 488 Not acceptable here
> CSeq: 3 INVITE
> Via: SIP/2.0/TLS
> 192.168.0.7:53084;branch=z9hG4bKc509aaac-a508-1910-9707-18037327ae65;received=192.168.0.7;rport=53084
> Server: Asterisk 1.8.12
> From: <sip:2310@192.168.1.151>;tag=fbe4a6ac-a508-1910-9704-18037327ae65
> Call-ID: f5e5a6ac-a508-1910-9704-18037327ae65@dbusacca
> Supported: replaces, timer
> To: <sip:600@192.168.1.151>;tag=as62e6f602
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
> INFO, PUBLISH
> Content-Length: 0
>
>
>
>
>
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