Thanks! Now all is OK! На ср, 4.09.2019 г. в 15:34 ч. Ben Newlin <ben.new...@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-boun...@lists.opensips.org on behalf of alexei.vasil...@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...@sip.test.dk SIP/2.0 > Via: SIP/2.0/TLS 172.16.1.29:4169;branch=z9hG4bK-gci2vl6fe7cz;rport > From: "Demo" <sip:2...@sip.test.dk>;tag=ncsplp1nvz > To: <sip:*7...@sip.test.dk> > Call-ID: 313536373539393535363232353137-eewp9wlm45rf > CSeq: 2 INVITE > Max-Forwards: 70 > User-Agent: snom320/8.7.5.44 > Contact: <sip:200@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...@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 > > _______________________________________________ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users >
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