[OpenSIPS-Users] Double SDP

[Dragomir Haralambiev]

Thanks!
Now all is OK!

На ср, 4.09.2019 г. в 15:34 ч. Ben Newlin <Ben.Newlin at genesys.com> написа:

> 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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>
>
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