[OpenSIPS-Users] Double SDP

[Ben Newlin]

If you don't want to have both in the second INVITE, you can try putting both rtpengine_offer calls in branch routes instead. I haven't worked with rtpengine, but with other messages changes like this if you place them in the branch route then they affect only the current branch; after failure the original message will be returned and you may then be able to add RTP/SAVP only.

Ben Newlin 

On 9/4/19, 8:27 AM, "Users on behalf of Alexey Vasilyev" <users-bounces at lists.opensips.org on behalf of alexei.vasilyev at gmail.com> wrote:

    This is absolutely normal. SDP can contain both RTP/AVP and RTP/SAVP. This is
    Invite from snom phone, for example:
    
    Sent to tls:135.42.212.82:5061 at Sep 4 14:19:18.641 (1383 bytes):
    
    INVITE sip:*7 at sip.test.dk SIP/2.0
    Via: SIP/2.0/TLS 172.16.1.29:4169;branch=z9hG4bK-gci2vl6fe7cz;rport
    From: "Demo" <sip:200 at sip.test.dk>;tag=ncsplp1nvz
    To: <sip:*7 at sip.test.dk>
    Call-ID: 313536373539393535363232353137-eewp9wlm45rf
    CSeq: 2 INVITE
    Max-Forwards: 70
    User-Agent: snom320/8.7.5.44
    Contact: <sip:200 at 172.16.1.29:4169;transport=tls>;reg-id=1
    X-Serialnumber: 000XXX
    P-Key-Flags: keys="3"
    Accept: application/sdp
    Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
    MESSAGE, INFO, UPDATE
    Allow-Events: talk, hold, refer, call-info
    Supported: timer, 100rel, replaces, from-change
    Session-Expires: 600
    Min-SE: 90
    Authorization: Digest
    username="200",realm="asterisk",nonce="7b2d56ec",uri="sip:*7 at sip.test.dk",response="7a9fe1f24a6f7585fb7323237a000167",algorithm=MD5
    Content-Type: application/sdp
    Content-Length: 476
    
    v=0
    o=root 558099897 558099897 IN IP4 172.16.1.29
    s=call
    c=IN IP4 172.16.1.29
    t=0 0
    m=audio 60812 RTP/SAVP 9 8 101
    a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:ysn2nTlXXXXXXAuZYcpOhf1g/h+oG
    a=rtpmap:9 G722/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=ptime:20
    a=sendrecv
    m=audio 60812 RTP/AVP 9 8 101
    a=rtpmap:9 G722/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=ptime:20
    a=sendrecv
    
    
    
    
    
    -----
    ---
    Alexey Vasilyev
    --
    Sent from: http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html
    
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