Displaying 20 results from an estimated 22 matches for "msid".
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msi
2014 Aug 22
0
Asterisk rejects sdp from webrtc client
...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 through the differences)
v=0
o=- 9046935681162021751 2 IN IP4 91.221.66.61
s=-
t=0 0
a=group:BUNDLE audio //
a=msid-semantic: WMS Kqg5QpXyqNeviT8qxUIRi8QNUaV7mUnFiDIF
m=audio 11076 RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 91.221.66.61
a=candidate:2999745851 1 udp 2122260223 192.168.56.1 52820 typ host
generation 0 //
a=candidate:2999745851 2 udp 2122260223 192.168.56.1 52820 typ host
generation 0 //
a=c...
2019 Dec 12
2
asterisk pjsip webrtc rtp to private IP
...ireshark i need decrypt traffic every call which is time
consuming. get debug 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 hos...
2014 Mar 14
0
sipML5, Ast12 and WebRTC: not acceptable here
...guage="en,fr,it"
Call-ID: 636a5d79-5fda-f79a-cc4b-9ba18d060edc
CSeq: 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 t...
2015 May 21
1
asterisk 13 webrtc
...oubango Telecom
v=0
o=mozilla...THIS_IS_SDPARTA-38.0.1 4294967295 0 IN IP4 127.0.0.1
s=Doubango Telecom - firefox
t=0 0
a=sendrecv
a=fingerprint:sha-256
A4:67:26:11:1F:1E:F2:8F:75:02:FE:69:2F:FC:FA:87:7A:2C:DA:86:6D:40:43:31:B7:4C:89:0B:15:44:00:56
a=group:BUNDLE sdparta_0
a=ice-options:trickle
a=msid-semantic:WMS *
m=audio 52438 UDP/TLS/RTP/SAVPF 109 9 0 8
c=IN IP4 2.2.2.2
a=candidate:0 1 UDP 2128609535 10.128.3.220 52438 typ host
a=candidate:5 1 UDP 2128543999 192.168.56.1 52439 typ host
a=candidate:0 2 UDP 2128609534 10.128.3.220 52440 typ host
a=candidate:5 2 UDP 2128543998 192.168.56.1 5244...
2014 Mar 26
0
Secure audio cannot be provided
...is error WARNING[31977][C-00000009]: chan_sip.c:10657 process_sdp: Can't provide secure audio requested in SDP offer
>From the sdp can anyone suggest why secure audio cannot be provided
????v=0
????o=- 6611325078116277019 2 IN IP4 127.0.0.1
????s=-
????t=0 0
????a=group:BUNDLE audio
????a=msid-semantic: WMS YxFi1hLhslP6PiA3D1xi2RxV5i1iATmDOz4l
????m=audio 34054 UDP/TLS/RTP/SAVPF 111 103 104 0 8 107 106 105 13 126
????c=IN IP4 10.1.1.2
????a=rtcp:34054 IN IP4 10.1.1.2
????a=candidate:53234734 1 udp 2113937151 10.1.1.2 34054 typ host generation 0
????a=candidate:53234734 2 udp 2113937151 1...
2019 May 10
4
Asterisk 13.26.0 webRTC: Asterisk not passing along video
...[May 10 10:45:24] Content-Type: application/sdp
[May 10 10:45:24] Content-Length: 5098
[May 10 10:45:24]
[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 192.168.1.11...
2015 Apr 28
0
hi list need your help
...ihs2e8.invalid;transport=ws;ob>
Content-Type: application/sdp
Session-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...
2019 Dec 12
2
asterisk pjsip webrtc rtp to private IP
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
2017 Oct 30
0
Asterisk 15.1.0 Now Available
...nnel hangup
(Reported by Matt Jordan)
* ASTERISK-27182 - bridge: Crash when mapping streams
(Reported by Joshua Colp)
* ASTERISK-27180 - channel: requester leaks joint_cap on
success.
(Reported by Corey Farrell)
* ASTERISK-27179 - res_pjsip_session: Handling of 'msid' is
incorrect
(Reported by Kevin Harwell)
* ASTERISK-27119 - res_pjsip: parse/add msid attribute when
webrtc is enabled
(Reported by Kevin Harwell)
* ASTERISK-27171 - Asterisk 15.0.0-Beta1 does not compile
(Reported by Ira Emus)
* ASTERISK-26659 - res_pjsi...
2019 Dec 12
2
asterisk pjsip webrtc rtp to private IP
...at 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=candid...
2015 Mar 23
2
PJSIP - Video Support for WebRTC
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
2023 Jun 28
1
SDP a=ice-ufrag & a=ice-pwd UNSUPPORTED OR FAILED
...[C-00000000] chan_sip.c: Processing media-level (audio) SDP
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid... UNSUPPORTED OR FAILED.
DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP
a=sendrecv... OK.
DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP
a=msid:4f4db37d-65ff-4f57-8c1c-b404f976c3fb
cc4a3d72-3e9d-4926-b57c-056b6e7a6d6c... UNSUPPORTED OR FAILED.
DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP
a=rtcp-mux... OK.
DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP
a=rtpmap:111 opus/48000/2... OK.
D...
2015 May 04
0
Asterisk proxying a REFER
.../sdp
> Session-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...
2016 Aug 09
2
Asterisk 11.23.0 on CentOS6 : how to get ICE support ?
...sipML5-v1.2016.03.04
[Aug 9 22:15:50] Organization: Doubango Telecom
[Aug 9 22:15:50]
[Aug 9 22:15:50] v=0
[Aug 9 22:15:50] o=- 9108976588890881000 2 IN IP4 127.0.0.1
[Aug 9 22:15:50] s=Doubango Telecom - chrome
[Aug 9 22:15:50] t=0 0
[Aug 9 22:15:50] a=group:BUNDLE audio
[Aug 9 22:15:50] a=msid-semantic: WMS BJSlrOtzPj6wzI3QugifY58Oi18zpEbkNsps
[Aug 9 22:15:50] m=audio 41178 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106
105 13 126
[Aug 9 22:15:50] c=IN IP4 178.119.146.190
[Aug 9 22:15:50] a=rtcp:42197 IN IP4 178.119.146.190
[Aug 9 22:15:50] a=candidate:1668076467 1 udp 2122260223 192.168.1...
2023 Jun 27
1
Get channel variables via ARI/AMI
I need to get hooked up with this class, I could have students doing
projects for homework :) Interested in RTCP?
j
On 6/26/23 7:45 PM, TTT wrote:
>
> I’m in training, so I have to demonstrate something SIP related. I
> figure it would be cool to hack a call, hanging it up while in
> progress from outside Asterisk. Doing so will demonstrate
> use/knowledge of ARI, AMI, SIP,
2016 Aug 11
2
Asterisk 11.23.0 on CentOS6 : how to get ICE support ?
...sipML5-v1.2016.03.04
[Aug 11 15:53:47] Organization: Doubango Telecom
[Aug 11 15:53:47]
[Aug 11 15:53:47] v=0
[Aug 11 15:53:47] o=- 5876454736929512000 2 IN IP4 127.0.0.1
[Aug 11 15:53:47] s=Doubango Telecom - chrome
[Aug 11 15:53:47] t=0 0
[Aug 11 15:53:47] a=group:BUNDLE audio
[Aug 11 15:53:47] a=msid-semantic: WMS kBwsfCPizGNiVjZS23dGoTNcUubDMMhxXrka
[Aug 11 15:53:47] m=audio 63897 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106
105 13 126
[Aug 11 15:53:47] c=IN IP4 178.119.146.190
[Aug 11 15:53:47] a=rtcp:63899 IN IP4 178.119.146.190
[Aug 11 15:53:47] a=candidate:2999745851 1 udp 2122260223 192.168.5...
2016 Aug 10
2
Asterisk 11.23.0 on CentOS6 : how to get ICE support ?
Hello
thank you for your answer.
I don't understand how there are many tutorials and examples on the web
where every time the outcome is a working setup. Very strange I feel now
after my personal experience with Asterisk 11 and webRTC.
You also say Asterisk 13. How about Asterisk 12 then ??
Kind regards.
On 10-08-16 21:53, Matt Fredrickson wrote:
> I don't see an ice-ufrag or
2020 Jun 08
0
pjsip extensions rings but call drop on answer
...9;asterisk2' and port ''.
[Jun 8 12:28:09] DEBUG[4180] res_rtp_asterisk.c: Setup RTCP on RTP
instance '0x7f05780618e0'
[Jun 8 12:28:09] DEBUG[4180] res_srtp.c: local_key64
2Rbo7TRiuRAnS0IYJeSn0ELEYAVnkOVCUwou7pxO len 40
[Jun 8 12:28:09] DEBUG[4180] res_pjsip_sdp_rtp.c: Stream msid:
0x7f0578077610 audio 23eb03ca-f0ee-406a-b7cd-5fb19fc33fa2
ddca7927-ff8d-45ab-a61f-9474f8b7a9df
[Jun 8 12:28:09] DEBUG[4180] res_pjsip_session.c: Method is INVITE
[Jun 8 12:28:09] DEBUG[4180] res_pjsip/pjsip_resolver.c: Performing
SIP DNS resolution of target '10.215.144.48'
[Jun 8 12:28...
2018 Oct 09
0
Asterisk 16.0.0 Now Available
...l hangup
(Reported by Matt Jordan)
* ASTERISK-27182 - bridge: Crash when mapping streams
(Reported by Joshua C. Colp)
* ASTERISK-27180 - channel: requester leaks joint_cap on
success.
(Reported by Corey Farrell)
* ASTERISK-27179 - res_pjsip_session: Handling of 'msid' is
incorrect
(Reported by Kevin Harwell)
* ASTERISK-27119 - res_pjsip: parse/add msid attribute when
webrtc is enabled
(Reported by Kevin Harwell)
* ASTERISK-27171 - Asterisk 15.0.0-Beta1 does not compile
(Reported by Ira Emus)
* ASTERISK-26659 - res_pjsi...
2017 Oct 03
0
Asterisk 15.0.0 Now Available
...led and "or" is not
available, external components (excluding pjsip) are not
installed
(Reported by Se??n C. McCord)
* ASTERISK-27200 - manager: hook event is not being raised
(Reported by Kevin Harwell)
* ASTERISK-27179 - res_pjsip_session: Handling of 'msid' is
incorrect
(Reported by Kevin Harwell)
* ASTERISK-27182 - bridge: Crash when mapping streams
(Reported by Joshua Colp)
* ASTERISK-27189 - Make --with-pjproject-bundled the default
for Asterisk 15
(Reported by George Joseph)
* ASTERISK-27180 - channel: r...