search for: savpf

...nts are successfully connected. But when I made a call from one of the users, the other user doen'st have call notification and for a while the calling process ended. I check the /var/log/asterisk/messages got the following log: [Dec 17 14:54:11] WARNING[11471][C-00000000] chan_sip.c: Received SAVPF profle in audio offer but AVPF is not enabled: audio 52760 RTP/SAVPF 103 104 0 8 107 106 105 13 126 [Dec 17 14:54:11] WARNING[11471][C-00000000] chan_sip.c: Received SAVPF profle in video offer but AVPF is not enabled: video 52760 RTP/SAVPF 100 101 102 [Dec 17 14:54:11] WARNING[11471][C-00000000] c...

Greetings, 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. 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...

I've seen the following appear in some tests with Asterisk 11.11: WARNING[3938][C-00000003]: chan_sip.c:10535 process_sdp: Rejecting secure audio stream without encryption details: audio 54908 UDP/TLS/RTP/SAVPF 109 0 8 101 Specifically, it always happens from a Firefox 24 host but it works without this error from another host running Firefox 26 I did a diff on the SDP and couldn't see anything obviously different except one thing: Firefox 24 only has host candidates for ICE (TURN support was only a...

...a known issue" or "no, and this should be reported to Mozilla", that would be very helpful for me as well. Here is the error I see in the Asterisk console after it successfully parses the SDP a lines: Rejecting secure audio stream without encryption details: audio 62583 UDP/TLS/RTP/SAVPF 109 0 8 101 Trying to put 'SIP/2.0 488' onto WS socket destined for www.xxx.yyy.zzz:5060 No compatible codecs for this SIP call. Here is the sip.conf info. I have tried various permutations of the dtls and encryption parameters with no luck. I do have openssl and srtp built into Asterisk...

...ebug from pjnat through asterisk is not possible because of technical reasons or nobody did it? in my case its strange that ice candidates are the same good call v=0 o=- 3669976329745317845 2 IN IP4 127.0.0.1 s=- t=0 0 a=msid-semantic: WMS EoNIdKcMZvWBLULGqGPJTDe12ujjFEemeapo m=audio 52421 RTP/SAVPF 8 0 101 c=IN IP4 10.2.152.36 a=rtcp:9 IN IP4 0.0.0.0 a=candidate:3607370648 1 udp 2122260223 10.2.152.36 52421 typ host generation 0 network-id 1 network-cost 10 a=candidate:2575820648 1 tcp 1518280447 10.2.152.36 9 typ host tcptype active generation 0 network-id 1 network-cost 10 bad call v=0...

...d something came up worth asking: While calling from a webrtc client to another (chrome, sip.js) Asterisk receives the following sdp and rejects it with 488 Not Acceptable. Why does this happen, what's wrong with the sdp? The second sdp body below is accepted instead. Both have rtp profile RTP/SAVPF, difference is that the second one was produced by rtpengine, first one came directly from the client. I defined my clients according to the sip.js guide: http://sipjs.com/guides/server-configuration/asterisk/ So this was rejected: (I marked the extra lines with '//' to ease looking throu...

...Expires: 90 Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS Supported: timer,ice,outbound User-Agent: JsSIP 0.6.26 Content-Length: 2554 v=0 o=- 4785391175048354014 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE audio video a=msid-semantic: WMS cC3clldcCIxZyWm8eHpKdycUakfANCZmV8Br m=audio 2313 RTP/SAVPF 111 103 104 9 0 8 106 105 13 126 c=IN IP4 192.168.88.26 a=rtcp:2313 IN IP4 192.168.88.26 a=candidate:97470069 1 udp 2122260223 192.168.88.26 2313 typ host generation 0 a=candidate:97470069 2 udp 2122260223 192.168.88.26 2313 typ host generation 0 a=candidate:1263319685 1 tcp 1518280447 192.168.88.2...

Hello list, before to lost my time, I'd like know if someone have a WebRTC working configuration on Asterisk 13.11.0 SIP or PJSIP channel. Thank you Regards

I'm using v11.11 I tried setting: force_avp=yes in a SIP peer in sip.conf and it seems to be ignored. The WebRTC client sends an INVITE with "RTP/SAVPF" and Asterisk is still sending back 183 and 200 responses with the UDP/TLS/RTP/SAVPF string Are there some limitations with this option or does it depend on any other settings? Is there any debugging I can enable to understand what is going wrong?

Hey i have an interesting topic to discuss here. The main goal here is to be able to make a video call between two WebRTC endpoints registered on asterisk 13 it is a feature that definitely asterisk 13 should support . the problems that i faced with this is the following and i hope i could get an advise here. asterisk 13 vanilla version has some issues marking the video packets this complain

...Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 1799 v=0 o=root 469858785 469858785 IN IP4 10.1.0.4 s=Asterisk PBX 11.21.0 c=IN IP4 10.1.0.4 b=CT:384 t=0 0 m=audio 14814 UDP/TLS/RTP/SAVPF 4 0 3 8 18 110 9 101 a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:110 speex/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=ice-ufrag:0e746cd50c88ce6e...

...This is interesting technology and I try to find out how to connect all the moving parts. Firefox: Neither sipml5 or jssip works with calls to asterisk, audio/video doesn't matter. WARNING[977][C-00000005] chan_sip.c: Rejecting secure audio stream without encryption details: audio 35684 RTP/SAVPF 109 0 8 101 --> Asterisk sends "SIP/2.0 488 Not acceptable here" Chrome: I've tried both sipml5 and jssip softphones and they both work. Even video + confbridge works with some minor quirks (lost connections sometimes, I guess plain old nat issues). Just relaying audio+video wit...

...,OPTIONS > Supported: timer,ice,outbound > User-Agent: JsSIP 0.6.26 > Content-Length: 2554 > > v=0 > o=- 4785391175048354014 2 IN IP4 127.0.0.1 > s=- > t=0 0 > a=group:BUNDLE audio video > a=msid-semantic: WMS cC3clldcCIxZyWm8eHpKdycUakfANCZmV8Br > m=audio 2313 RTP/SAVPF 111 103 104 9 0 8 106 105 13 126 > c=IN IP4 192.168.88.26 > a=rtcp:2313 IN IP4 192.168.88.26 > a=candidate:97470069 1 udp 2122260223 192.168.88.26 2313 typ host > generation 0 > a=candidate:97470069 2 udp 2122260223 192.168.88.26 2313 typ host > generation 0 > a=candidate:12633...

hi, i have following topology PSTN - Asterisk ---- internet -----  router - jssip client (wss) Asterisk 13.29.1 on public IP, chan_pjsip for wss, chan_sip/udp for SIP connection to PSTN router - public IP/private IP (NAT) jssip client - private IP - sip over websocket to Asterisk PJSIP ~30% of calls has problem with no audio. reason is that Asterisk is sending RTP to private IP of jssip

...5:24] [May 10 10:45:24] v=0 [May 10 10:45:24] o=- 6075323372920596423 2 IN IP4 127.0.0.1 [May 10 10:45:24] s=- [May 10 10:45:24] t=0 0 [May 10 10:45:24] a=group:BUNDLE audio video [May 10 10:45:24] a=msid-semantic: WMS I46iog3EpKvlzvX9g0MMsh3ON7hT9qtZwZ4E [May 10 10:45:24] m=audio 34197 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126 [May 10 10:45:24] c=IN IP4 99.99.255.55 [May 10 10:45:24] a=rtcp:9 IN IP4 0.0.0.0 [May 10 10:45:24] a=candidate:2395300328 1 udp 2122260223 192.168.1.110 34197 typ host generation 0 network-id 1 network-cost 10 [May 10 10:45:24] a=candidate:260925276 1...

...Seq: 38718 INVITE Content-Type: application/sdp Content-Length: 1827 Max-Forwards: 70 User-Agent: IM-client/OMA1.0 sipML5 v=0 o=- 365893986064703740 2 IN IP4 127.0.0.1 s=Doubango Telecom - chrome t=0 0 a=group:BUNDLE audio a=msid-semantic: WMS dXVhxyOSxULu3iClZayhTeEBzH2voboiJJ28 m=audio 37874 RTP/SAVPF 111 103 104 0 8 106 105 13 126 c=IN IP4 85.0.XXX.XXX a=rtcp:37874 IN IP4 85.0.XXX.XXX a=candidate:296123718 1 udp 2113937151 10.10.5.106 63858 typ host generation 0 a=candidate:296123718 2 udp 2113937151 10.10.5.106 63858 typ host generation 0 a=candidate:3103388307 1 udp 1845501695 85.0.XXX.XXX...

...ded run chan_sip for old hard phones and wss with chan_pjsip only for webrtc. this is possible with patch from https://issues.asterisk.org/jira/browse/ASTERISK-24106 chan_sip is not usable for webrtc because of https://issues.asterisk.org/jira/browse/ASTERISK-24602 another problem arise with RTP/SAVPF negotiation this can be solved with patch for Asterisk from https://issues.asterisk.org/jira/browse/ASTERISK-24602 and for pjsip http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/2015-September/018607.html i hope this info helps what is your experience with WebRTC? See you at WebRTC Expo Pa...

...did it? > > > in my case its strange that ice candidates are the same > > good call > > v=0 > o=- 3669976329745317845 2 IN IP4 127.0.0.1 > s=- > t=0 0 > a=msid-semantic: WMS EoNIdKcMZvWBLULGqGPJTDe12ujjFEemeapo > m=audio 52421 RTP/SAVPF 8 0 101 > c=IN IP4 10.2.152.36 > a=rtcp:9 IN IP4 0.0.0.0 > a=candidate:3607370648 1 udp 2122260223 10.2.152.36 52421 typ host > generation 0 network-id 1 network-cost 10 > a=candidate:2575820648 1 tcp 1518280447 10.2.152.36 9 typ host > tcptype active gener...

I've just done a test with a WebRTC client connecting to the repro proxy with the SIP messages relayed over TCP to Asterisk Asterisk successfully answers the call using SAVPF, SRTP and ICE. The client is greeted by the demo This was tested in the Asterisk 11 environment described in my earlier email about SRTP build issues on the asterisk-users list. This is quite useful because it proves that Asterisk doesn't have to be exposed as the HTTP WebSocket server: all...

...k_utils_log_errortsk_utils.js?svn=224:128 2. tmedia_session_jsep01.onIceCandidate tmedia_session_jsep.js?svn=224:677 3. o_pc.onicecandidate and if I do call form sipml5 client to blink the i get *Rejecting secure audio stream without encryption details: audio 59476 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126* I had enabled dtls-srtp in asterisk (with or without this getting the same error) I am struggling in this from a long time please help me -- Regards Sameer Rathod 8109413462 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http:...