<div dir="ltr">Hi,<div><br></div><div>I got ahead with my setup, this post helped me much: <a href="http://forums.digium.com/viewtopic.php?f=1&t=90167&sid=66fdf8cc4be5d955ba584e989a23442f">http://forums.digium.com/viewtopic.php?f=1&t=90167&sid=66fdf8cc4be5d955ba584e989a23442f</a></div>
<div><br></div><div>At least the avpf setting had to be removed from sip.conf and put in the realtime db table, defined per client. I left the encryption setting in sip.conf. I had some problems calling from SIP client to another, then had to define avpf=no for those clients. Personally I don't like to use different settings to different clients, is there a way around this? </div>
<div><br></div><div>With this setup I can make calls between SIP clients but not ws clients. My client (now I use sip.js) fails to parse the sdp - including the apparently correct rtp profile UDP/TLS/RTP/SAVPF - and sends back 488, which makes the call fail. I'd like to hear opinions from you guys which would be the correct place to handle this? My setup has Asterisk Kamailio realtime integration, and I use dispatcher in Kamailio to route calls to Asterisk. Kamailio sounds like the logical place, but I'd rather find a way to not change the rtp profile along the way, at least until the clients can support that one.</div>
<div><br></div><div>cheers,</div><div>Olli</div><div><br></div><div><br></div><div><br></div></div><div class="gmail_extra"><br><br><div class="gmail_quote">2014-07-26 12:58 GMT+03:00 Olli Heiskanen <span dir="ltr"><<a href="mailto:ohjelmistoarkkitehti@gmail.com" target="_blank">ohjelmistoarkkitehti@gmail.com</a>></span>:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div><div><br></div><div>Greetings,</div><div><br></div><div>I've noticed a problem that might originate from my Asterisk configuration, could use a hand in sorting it out. Problem is a 488 response from Asterisk whenever it gets RTP/SAVPF profile in the SDP.</div>
<div><br></div><div>My current setup has Asterisk Kamailio realtime integration, and Kamailio uses dispatcher to route calls for Asterisk to handle. Now I have only one Asterisk, on the same machine as Kamailio. The version is 11.10.2. With Kamailio I use rtpengine, which affects SDP descriptions when 488 response is received. </div>
<div><br></div><div>My goal is to enable two websocket clients using Chrome to call each other, using Kamailio as outbound proxy. Kamailio routes signaling to Asterisk, and then back to clients. Currently the problem is RTP, when INVITE is received from client A to Kamailio, it is relayed to Asterisk. Asterisk responds with 488 Not Acceptable here and the cli says: </div>
<div><br></div><div> NOTICE[11642][C-00000006]: chan_sip.c:10124 process_sdp: Received SAVPF profle in audio offer but AVPF is not enabled, enabling: audio 30212 RTP/SAVPF 111 103 104 0 8 106 105 13 126</div><div> WARNING[11642][C-00000006]: chan_sip.c:10509 process_sdp: Rejecting secure audio stream without encryption details: audio 30212 RTP/SAVPF 111 103 104 0 8 106 105 13 126</div>
<div><br></div><div><br></div><div>Strange thing is, I don't know why Asterisk says AVPF is not enabled. The warning about rejecting the audio stream must be behind the 488 response but I didn't find any answers that would solve my case so I must turn to you guys. In my sip.conf I have savpf=yes, but is there something else I need to enable or change in the configs or change my peer configurations?</div>
<div><br></div><div>I'm not sure if this is relevant but I checked that Asterisk was successfully compiled with res_srtp module.</div><div><br></div><div>Here's my sip.conf contents: </div><div> </div><div>bindport = 5070<span style="white-space:pre-wrap"> </span>; using this since Kamailio is at 5060</div>
<div>bindaddr = PU.BL.IC.IP</div><div>tcpenable = yes ;no</div><div>limitonpeers = yes</div><div>rtcachefriends = yes ; for realtime</div><div>rtupdate=yes</div><div>tos_sip=cs3</div><div>tos_audio=ef</div><div>useragent=MyAsterisk</div>
<div>realm = <a href="http://myrealm.com" target="_blank">myrealm.com</a></div><div><br></div><div>autodomain=no</div><div>domain=PU.BL.IC.IP</div><div>domain=<a href="http://testers.com" target="_blank">testers.com</a></div>
<div><br></div><div>allowexternaldomains=no</div>
<div>allowguest=no</div><div>avpf=yes</div><div>encryption=yes</div><div><br></div><div>transport=ws,udp</div><div>icesupport=yes</div><div>srvlookup=yes</div><div><br></div><div><br></div><div>And here's an example of a ws client in my realtime peer table:</div>
<div><br></div><div> id: 4</div><div> name: 660</div><div> ipaddr: PU.BL.IC.IP</div><div> port: 5060</div><div> regseconds: 1406368294</div><div> defaultuser: 660</div>
<div> fullcontact: sip:660@PU.BL.IC.IP:5060</div><div> regserver:</div><div> useragent:</div><div> lastms: 0</div><div> host: dynamic</div><div> type: friend</div>
<div> context: default</div><div> deny: <a href="http://0.0.0.0/0.0.0.0" target="_blank">0.0.0.0/0.0.0.0</a></div><div> permit: PU.BL.IC.IP</div><div> secret: NULL</div><div> md5secret: NULL</div>
<div> remotesecret: NULL</div><div> transport: NULL</div><div> dtmfmode: NULL</div><div> directmedia: NULL</div><div> nat: force_rport,comedia</div><div> callgroup: NULL</div>
<div> pickupgroup: NULL</div><div> language: NULL</div><div> disallow: NULL</div><div> allow: NULL</div><div> insecure: NULL</div><div> trustrpid: NULL</div><div> progressinband: NULL</div>
<div> promiscredir: NULL</div><div> useclientcode: NULL</div><div> accountcode: NULL</div><div> setvar: NULL</div><div> callerid: NULL</div><div> amaflags: NULL</div><div> callcounter: NULL</div>
<div> busylevel: NULL</div><div> allowoverlap: NULL</div><div> allowsubscribe: NULL</div><div> videosupport: NULL</div><div> maxcallbitrate: NULL</div><div> rfc2833compensate: NULL</div><div> mailbox: NULL</div>
<div> session-timers: NULL</div><div> session-expires: NULL</div><div> session-minse: NULL</div><div> session-refresher: NULL</div><div>t38pt_usertpsource: NULL</div><div> regexten: NULL</div><div> fromdomain: <a href="http://testers.com" target="_blank">testers.com</a></div>
<div> fromuser: 660</div><div> qualify: NULL</div><div> defaultip: NULL</div><div> rtptimeout: NULL</div><div> rtpholdtimeout: NULL</div><div> sendrpid: NULL</div><div> outboundproxy: PU.BL.IC.IP</div>
<div> timert1: NULL</div><div> timerb: NULL</div><div> qualifyfreq: NULL</div><div> constantssrc: NULL</div><div> contactpermit: NULL</div><div> contactdeny: NULL</div><div> usereqphone: NULL</div>
<div> textsupport: NULL</div><div> faxdetect: NULL</div><div> buggymwi: NULL</div><div> auth: NULL</div><div> fullname: NULL</div><div> trunkname: NULL</div><div> cid_number: NULL</div>
<div> callingpres: NULL</div><div> mohinterpret: NULL</div><div> mohsuggest: NULL</div><div> parkinglot: NULL</div><div> hasvoicemail: NULL</div><div> subscribemwi: NULL</div><div> vmexten: NULL</div>
<div> autoframing: NULL</div><div> rtpkeepalive: NULL</div><div> call-limit: NULL</div><div> g726nonstandard: NULL</div><div> ignoresdpversion: NULL</div><div> allowtransfer: NULL</div><div> dynamic: NULL</div>
<div> path: NULL</div><div> supportpath: NULL</div><div> sippasswd: my-md5-pwd</div><div> rpid: NULL</div><div> domain: <a href="http://testers.com" target="_blank">testers.com</a></div>
<div> sippasswd2: NULL</div><div><br></div><div><br></div><div>I'd greatly appreciate help on this!</div><div><br></div><div>cheers,</div><div>Olli</div></div></div>
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