3 Apr
2014
3 Apr
'14
4:22 p.m.
Hi List,
Can anyone help me understand why this is getting rejected
Please note the specific message further dow the log.
"Failed to parse SessionDescription. Failed to parse audio codecs correctly"
This is on Chrome.
On Firefox There is a further message in the console
"Could not negotiate answer SDP; cause = ERR | SDP Parsing Error: Warning: Transport
protocol type unsupported (UDP/TLS/RTP/SAVPF). | SDP Parsing Error: Invalid port format
(17296) specified for transport protocol (Unsupported), parse failed."
JsSIP | RTC SESSION | got local media stream jssip-0.3.0.js:3414
JsSIP | RTC SESSION | ICE candidate received: a=candidate:642192370 1 udp 2113937151
10.10.10.63 65223 typ host generation 0
jssip-0.3.0.js:3369
JsSIP | RTC SESSION | ICE candidate received: a=candidate:642192370 2 udp 2113937151
10.10.10.63 65223 typ host generation 0
jssip-0.3.0.js:3369
JsSIP | RTC SESSION | ICE candidate received: a=candidate:2999745851 1 udp 2113937151
192.168.56.1 65224 typ host generation 0
jssip-0.3.0.js:3369
JsSIP | RTC SESSION | ICE candidate received: a=candidate:2999745851 2 udp 2113937151
192.168.56.1 65224 typ host generation 0
jssip-0.3.0.js:3369
JsSIP | RTC SESSION | ICE candidate received: a=candidate:1757736706 1 tcp 1509957375
10.10.10.63 0 typ host generation 0
jssip-0.3.0.js:3369
JsSIP | RTC SESSION | ICE candidate received: a=candidate:1757736706 2 tcp 1509957375
10.10.10.63 0 typ host generation 0
jssip-0.3.0.js:3369
JsSIP | RTC SESSION | ICE candidate received: a=candidate:4233069003 1 tcp 1509957375
192.168.56.1 0 typ host generation 0
jssip-0.3.0.js:3369
JsSIP | RTC SESSION | ICE candidate received: a=candidate:4233069003 2 tcp 1509957375
192.168.56.1 0 typ host generation 0
jssip-0.3.0.js:3369
JsSIP | TRANSPORT | sending WebSocket message:
INVITE sip:9822@10.1.1.101 SIP/2.0
Via: SIP/2.0/WSS flogvr403sb2.invalid;branch=z9hG4bK9149581
Max-Forwards: 69
To: <sip:9822@10.1.1.101>
From: <sip:webrtc@10.10.10.48>;tag=6tmeble9ov
Call-ID: 43oclsi0sva6n347bk5c
CSeq: 7435 INVITE
Contact: <sip:ce5egl03@flogvr403sb2.invalid;transport=ws;ob>
Allow: ACK,CANCEL,BYE,OPTIONS,INVITE
Content-Type: application/sdp
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.0
Content-Length: 1744
v=0
o=- 3746191339358890844 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS TNUolHZksseiQbV1o2j8kmZGxOkOjYsZYXh8
m=audio 65223 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 10.10.10.63
a=rtcp:65223 IN IP4 10.10.10.63
a=candidate:642192370 1 udp 2113937151 10.10.10.63 65223 typ host generation 0
a=candidate:642192370 2 udp 2113937151 10.10.10.63 65223 typ host generation 0
a=candidate:2999745851 1 udp 2113937151 192.168.56.1 65224 typ host generation 0
a=candidate:2999745851 2 udp 2113937151 192.168.56.1 65224 typ host generation 0
a=candidate:1757736706 1 tcp 1509957375 10.10.10.63 0 typ host generation 0
a=candidate:1757736706 2 tcp 1509957375 10.10.10.63 0 typ host generation 0
a=candidate:4233069003 1 tcp 1509957375 192.168.56.1 0 typ host generation 0
a=candidate:4233069003 2 tcp 1509957375 192.168.56.1 0 typ host generation 0
a=ice-ufrag:Dgp8HIJdmr1lFPCQ
a=ice-pwd:2yYxerrscdbTQhr0vbCTiju9
a=ice-options:google-ice
a=fingerprint:sha-256
C8:E9:57:CB:85:63:F7:C5:FC:15:3D:8B:A8:10:94:F4:C9:BB:86:48:E3:EE:A0:5E:FA:42:14:55:6F:68:3F:B7
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
a=ssrc:3445528109 cname:QbLF+sVLqHbEqUxY
a=ssrc:3445528109 msid:TNUolHZksseiQbV1o2j8kmZGxOkOjYsZYXh8
768e2e47-bc86-473d-bc2c-6e2340ace772
a=ssrc:3445528109 mslabel:TNUolHZksseiQbV1o2j8kmZGxOkOjYsZYXh8
a=ssrc:3445528109 label:768e2e47-bc86-473d-bc2c-6e2340ace772
jssip-0.3.0.js:519
JsSIP | TRANSPORT | received WebSocket text message:
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/WSS
flogvr403sb2.invalid;branch=z9hG4bK9149581;rport=56527;received=10.10.10.63
To: <sip:9822@10.1.1.101>
From: <sip:webrtc@10.10.10.48>;tag=6tmeble9ov
Call-ID: 43oclsi0sva6n347bk5c
CSeq: 7435 INVITE
Server: DXI WebRTC
Content-Length: 0
Warning: 392 10.10.10.48:6443 "Noisy feedback tells: pid=23455
req_src_ip=10.10.10.63 req_src_port=56527 in_uri=sip:9822@10.1.1.101
out_uri=sip:9822@10.10.10.111:5443 via_cnt==1"
jssip-0.3.0.js:670
JsSIP | TRANSPORT | received WebSocket text message:
SIP/2.0 200 OK
Via: SIP/2.0/WSS
flogvr403sb2.invalid;rport=56527;received=10.10.10.63;branch=z9hG4bK9149581
From: <sip:webrtc@10.10.10.48>;tag=6tmeble9ov
To: <sip:9822@10.1.1.101>;tag=as06b3db08
Call-ID: 43oclsi0sva6n347bk5c
CSeq: 7435 INVITE
Server: Easycall
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH,
MESSAGE
Supported: replaces, timer
Contact: <sip:9822@10.10.10.111:5443;transport=TLS>
Content-Type: application/sdp
Content-Length: 801
v=0
o=root 431209641 431209641 IN IP4 10.10.10.111
s=Asterisk PBX 12.2.0-rc1
c=IN IP4 10.10.10.111
t=0 0
m=audio 30490 UDP/TLS/RTP/SAVPF 0 126
a=rtpmap:0 PCMU/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=maxptime:150
a=ice-ufrag:5236e84b43b5d10c117e8ead0a340138
a=ice-pwd:51005fde4e6e9d3f1879fbbc15e0f092
a=candidate:Ha1f026f 1 UDP 2130706431 10.10.10.111 30490 typ host
a=candidate:S5bec7504 1 UDP 1694498815 91.236.117.4 30490 typ srflx
a=candidate:Ha1f026f 2 UDP 2130706430 10.10.10.111 30491 typ host
a=candidate:S5bec7504 2 UDP 1694498814 91.236.117.4 30492 typ srflx
a=connection:new
a=setup:active
a=fingerprint:SHA-256
13:BB:CF:88:C4:75:9B:F0:DA:36:0A:6D:5D:37:C9:26:6B:3C:82:3E:F6:92:AE:A7:AE:CF:FF:78:F5:86:D9:E8
a=sendrecv
jssip-0.3.0.js:670
Failed to parse SessionDescription. Failed to parse audio codecs correctly.
jssip-0.3.0.js:4512
JsSIP | DIALOG | new UAC dialog created with status CONFIRMED jssip-0.3.0.js:2523
JsSIP | TRANSPORT | sending WebSocket message:
ACK sip:9822@10.10.10.111:5443;transport=tls SIP/2.0
Via: SIP/2.0/WSS flogvr403sb2.invalid;branch=z9hG4bK431640
Max-Forwards: 69
To: <sip:9822@10.1.1.101>;tag=as06b3db08
From: <sip:webrtc@10.10.10.48>;tag=6tmeble9ov
Call-ID: 43oclsi0sva6n347bk5c
CSeq: 7435 ACK
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.0
Content-Length: 0
jssip-0.3.0.js:519
JsSIP | TRANSPORT | sending WebSocket message:
BYE sip:9822@10.10.10.111:5443;transport=tls SIP/2.0
Via: SIP/2.0/WSS flogvr403sb2.invalid;branch=z9hG4bK7689766
Max-Forwards: 69
To: <sip:9822@10.1.1.101>;tag=as06b3db08
From: <sip:webrtc@10.10.10.48>;tag=6tmeble9ov
Call-ID: 43oclsi0sva6n347bk5c
CSeq: 7436 BYE
Reason: SIP ;cause=488; text="Not Acceptable Here"
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.0
Content-Length: 0
jssip-0.3.0.js:519
JsSIP | RTC SESSION | closing INVITE session 43oclsi0sva6n347bk5c6tmeble9ov
jssip-0.3.0.js:4193
JsSIP | RTC SESSION | closing PeerConnection jssip-0.3.0.js:3392
JsSIP | DIALOG | dialog 43oclsi0sva6n347bk5c6tmeble9ovas06b3db08 deleted
jssip-0.3.0.js:2543
JsSIP | EVENT EMITTER | emitting event failed jssip-0.3.0.js:187
JsSIP | TRANSPORT | received WebSocket text message:
SIP/2.0 200 OK
Via: SIP/2.0/WSS
flogvr403sb2.invalid;rport=56527;received=10.10.10.63;branch=z9hG4bK7689766
From: <sip:webrtc@10.10.10.48>;tag=6tmeble9ov
To: <sip:9822@10.1.1.101>;tag=as06b3db08
Call-ID: 43oclsi0sva6n347bk5c
CSeq: 7436 BYE
Server: Easycall
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH,
MESSAGE
Supported: replaces, timer
Content-Length: 0
3 Apr
3 Apr
6:11 p.m.
My guess would be that it's due to a discrepancy between WebRTC and RFC
5764. WebRTC uses a protocol string of "RTP/SAVPF", while the RFC says
that "UDP/TLS/RTP/SAVPF" shall be used. You can try an SDP rewrite
operation to substitute one for the other. Or teach your non-RTC client
to use a different protocol string.
cheers
On 04/03/14 10:22, jaflong jaflong wrote:
Hi List,
Can anyone help me understand why this is getting rejected
Please note the specific message further dow the log.
"Failed to parse SessionDescription. Failed to parse audio codecs correctly"
This is on Chrome.
On Firefox There is a further message in the console
"Could not negotiate answer SDP; cause = ERR | SDP Parsing Error: Warning:
Transport protocol type unsupported (UDP/TLS/RTP/SAVPF). | SDP Parsing Error: Invalid
port format (17296) specified for transport protocol (Unsupported), parse failed."
JsSIP | RTC SESSION | got local media stream jssip-0.3.0.js:3414
JsSIP | RTC SESSION | ICE candidate received: a=candidate:642192370 1 udp 2113937151
10.10.10.63 65223 typ host generation 0
jssip-0.3.0.js:3369
JsSIP | RTC SESSION | ICE candidate received: a=candidate:642192370 2 udp 2113937151
10.10.10.63 65223 typ host generation 0
jssip-0.3.0.js:3369
JsSIP | RTC SESSION | ICE candidate received: a=candidate:2999745851 1 udp 2113937151
192.168.56.1 65224 typ host generation 0
jssip-0.3.0.js:3369
JsSIP | RTC SESSION | ICE candidate received: a=candidate:2999745851 2 udp 2113937151
192.168.56.1 65224 typ host generation 0
jssip-0.3.0.js:3369
JsSIP | RTC SESSION | ICE candidate received: a=candidate:1757736706 1 tcp 1509957375
10.10.10.63 0 typ host generation 0
jssip-0.3.0.js:3369
JsSIP | RTC SESSION | ICE candidate received: a=candidate:1757736706 2 tcp 1509957375
10.10.10.63 0 typ host generation 0
jssip-0.3.0.js:3369
JsSIP | RTC SESSION | ICE candidate received: a=candidate:4233069003 1 tcp 1509957375
192.168.56.1 0 typ host generation 0
jssip-0.3.0.js:3369
JsSIP | RTC SESSION | ICE candidate received: a=candidate:4233069003 2 tcp 1509957375
192.168.56.1 0 typ host generation 0
jssip-0.3.0.js:3369
JsSIP | TRANSPORT | sending WebSocket message:
INVITE sip:9822@10.1.1.101 SIP/2.0
Via: SIP/2.0/WSS flogvr403sb2.invalid;branch=z9hG4bK9149581
Max-Forwards: 69
To: <sip:9822@10.1.1.101>
From: <sip:webrtc@10.10.10.48>;tag=6tmeble9ov
Call-ID: 43oclsi0sva6n347bk5c
CSeq: 7435 INVITE
Contact: <sip:ce5egl03@flogvr403sb2.invalid;transport=ws;ob>
Allow: ACK,CANCEL,BYE,OPTIONS,INVITE
Content-Type: application/sdp
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.0
Content-Length: 1744
v=0
o=- 3746191339358890844 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS TNUolHZksseiQbV1o2j8kmZGxOkOjYsZYXh8
m=audio 65223 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 10.10.10.63
a=rtcp:65223 IN IP4 10.10.10.63
a=candidate:642192370 1 udp 2113937151 10.10.10.63 65223 typ host generation 0
a=candidate:642192370 2 udp 2113937151 10.10.10.63 65223 typ host generation 0
a=candidate:2999745851 1 udp 2113937151 192.168.56.1 65224 typ host generation 0
a=candidate:2999745851 2 udp 2113937151 192.168.56.1 65224 typ host generation 0
a=candidate:1757736706 1 tcp 1509957375 10.10.10.63 0 typ host generation 0
a=candidate:1757736706 2 tcp 1509957375 10.10.10.63 0 typ host generation 0
a=candidate:4233069003 1 tcp 1509957375 192.168.56.1 0 typ host generation 0
a=candidate:4233069003 2 tcp 1509957375 192.168.56.1 0 typ host generation 0
a=ice-ufrag:Dgp8HIJdmr1lFPCQ
a=ice-pwd:2yYxerrscdbTQhr0vbCTiju9
a=ice-options:google-ice
a=fingerprint:sha-256
C8:E9:57:CB:85:63:F7:C5:FC:15:3D:8B:A8:10:94:F4:C9:BB:86:48:E3:EE:A0:5E:FA:42:14:55:6F:68:3F:B7
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
a=ssrc:3445528109 cname:QbLF+sVLqHbEqUxY
a=ssrc:3445528109 msid:TNUolHZksseiQbV1o2j8kmZGxOkOjYsZYXh8
768e2e47-bc86-473d-bc2c-6e2340ace772
a=ssrc:3445528109 mslabel:TNUolHZksseiQbV1o2j8kmZGxOkOjYsZYXh8
a=ssrc:3445528109 label:768e2e47-bc86-473d-bc2c-6e2340ace772
jssip-0.3.0.js:519
JsSIP | TRANSPORT | received WebSocket text message:
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/WSS
flogvr403sb2.invalid;branch=z9hG4bK9149581;rport=56527;received=10.10.10.63
To: <sip:9822@10.1.1.101>
From: <sip:webrtc@10.10.10.48>;tag=6tmeble9ov
Call-ID: 43oclsi0sva6n347bk5c
CSeq: 7435 INVITE
Server: DXI WebRTC
Content-Length: 0
Warning: 392 10.10.10.48:6443 "Noisy feedback tells: pid=23455
req_src_ip=10.10.10.63 req_src_port=56527 in_uri=sip:9822@10.1.1.101
out_uri=sip:9822@10.10.10.111:5443 via_cnt==1"
jssip-0.3.0.js:670
JsSIP | TRANSPORT | received WebSocket text message:
SIP/2.0 200 OK
Via: SIP/2.0/WSS
flogvr403sb2.invalid;rport=56527;received=10.10.10.63;branch=z9hG4bK9149581
From: <sip:webrtc@10.10.10.48>;tag=6tmeble9ov
To: <sip:9822@10.1.1.101>;tag=as06b3db08
Call-ID: 43oclsi0sva6n347bk5c
CSeq: 7435 INVITE
Server: Easycall
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH,
MESSAGE
Supported: replaces, timer
Contact: <sip:9822@10.10.10.111:5443;transport=TLS>
Content-Type: application/sdp
Content-Length: 801
v=0
o=root 431209641 431209641 IN IP4 10.10.10.111
s=Asterisk PBX 12.2.0-rc1
c=IN IP4 10.10.10.111
t=0 0
m=audio 30490 UDP/TLS/RTP/SAVPF 0 126
a=rtpmap:0 PCMU/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=maxptime:150
a=ice-ufrag:5236e84b43b5d10c117e8ead0a340138
a=ice-pwd:51005fde4e6e9d3f1879fbbc15e0f092
a=candidate:Ha1f026f 1 UDP 2130706431 10.10.10.111 30490 typ host
a=candidate:S5bec7504 1 UDP 1694498815 91.236.117.4 30490 typ srflx
a=candidate:Ha1f026f 2 UDP 2130706430 10.10.10.111 30491 typ host
a=candidate:S5bec7504 2 UDP 1694498814 91.236.117.4 30492 typ srflx
a=connection:new
a=setup:active
a=fingerprint:SHA-256
13:BB:CF:88:C4:75:9B:F0:DA:36:0A:6D:5D:37:C9:26:6B:3C:82:3E:F6:92:AE:A7:AE:CF:FF:78:F5:86:D9:E8
a=sendrecv
jssip-0.3.0.js:670
Failed to parse SessionDescription. Failed to parse audio codecs correctly.
jssip-0.3.0.js:4512
JsSIP | DIALOG | new UAC dialog created with status CONFIRMED jssip-0.3.0.js:2523
JsSIP | TRANSPORT | sending WebSocket message:
ACK sip:9822@10.10.10.111:5443;transport=tls SIP/2.0
Via: SIP/2.0/WSS flogvr403sb2.invalid;branch=z9hG4bK431640
Max-Forwards: 69
To: <sip:9822@10.1.1.101>;tag=as06b3db08
From: <sip:webrtc@10.10.10.48>;tag=6tmeble9ov
Call-ID: 43oclsi0sva6n347bk5c
CSeq: 7435 ACK
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.0
Content-Length: 0
jssip-0.3.0.js:519
JsSIP | TRANSPORT | sending WebSocket message:
BYE sip:9822@10.10.10.111:5443;transport=tls SIP/2.0
Via: SIP/2.0/WSS flogvr403sb2.invalid;branch=z9hG4bK7689766
Max-Forwards: 69
To: <sip:9822@10.1.1.101>;tag=as06b3db08
From: <sip:webrtc@10.10.10.48>;tag=6tmeble9ov
Call-ID: 43oclsi0sva6n347bk5c
CSeq: 7436 BYE
Reason: SIP ;cause=488; text="Not Acceptable Here"
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.0
Content-Length: 0
jssip-0.3.0.js:519
JsSIP | RTC SESSION | closing INVITE session 43oclsi0sva6n347bk5c6tmeble9ov
jssip-0.3.0.js:4193
JsSIP | RTC SESSION | closing PeerConnection jssip-0.3.0.js:3392
JsSIP | DIALOG | dialog 43oclsi0sva6n347bk5c6tmeble9ovas06b3db08 deleted
jssip-0.3.0.js:2543
JsSIP | EVENT EMITTER | emitting event failed jssip-0.3.0.js:187
JsSIP | TRANSPORT | received WebSocket text message:
SIP/2.0 200 OK
Via: SIP/2.0/WSS
flogvr403sb2.invalid;rport=56527;received=10.10.10.63;branch=z9hG4bK7689766
From: <sip:webrtc@10.10.10.48>;tag=6tmeble9ov
To: <sip:9822@10.1.1.101>;tag=as06b3db08
Call-ID: 43oclsi0sva6n347bk5c
CSeq: 7436 BYE
Server: Easycall
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH,
MESSAGE
Supported: replaces, timer
Content-Length: 0
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
4 Apr
4 Apr
8:18 a.m.
Hallo,
my guess is the audio codec opus
asterisk can NOT do transcoding from opus to pcmu.
The opus codec in asterisk is (just) a path through codec.
your trace right at the end:
!!! Failed to parse SessionDescription. Failed to parse audio codecs correctly !!!
Regards
Rainer
Am 03.04.2014 18:11, schrieb Richard Fuchs:
My guess would be that it's due to a discrepancy
between WebRTC and RFC
5764. WebRTC uses a protocol string of "RTP/SAVPF", while the RFC says
that "UDP/TLS/RTP/SAVPF" shall be used. You can try an SDP rewrite
operation to substitute one for the other. Or teach your non-RTC client
to use a different protocol string.
cheers
On 04/03/14 10:22, jaflong jaflong wrote:
--
*Rainer Piper*
NOC - +49 (0)228 97167161 <callto:004922897167161> - sip.soho-piper.de
NOC - +49 (0)2247 9064188 <callto:004922479064188> - sip.tele33.de -
sip.tefonix.de - D293
Hi List,
Can anyone help me understand why this is getting rejected
Please note the specific message further dow the log.
"Failed to parse SessionDescription. Failed to parse audio codecs correctly"
This is on Chrome.
On Firefox There is a further message in the console
"Could not negotiate answer SDP; cause = ERR | SDP Parsing Error: Warning:
Transport protocol type unsupported (UDP/TLS/RTP/SAVPF). | SDP Parsing Error: Invalid
port format (17296) specified for transport protocol (Unsupported), parse failed."
JsSIP | RTC SESSION | got local media stream jssip-0.3.0.js:3414
JsSIP | RTC SESSION | ICE candidate received: a=candidate:642192370 1 udp 2113937151
10.10.10.63 65223 typ host generation 0
jssip-0.3.0.js:3369
JsSIP | RTC SESSION | ICE candidate received: a=candidate:642192370 2 udp 2113937151
10.10.10.63 65223 typ host generation 0
jssip-0.3.0.js:3369
JsSIP | RTC SESSION | ICE candidate received: a=candidate:2999745851 1 udp 2113937151
192.168.56.1 65224 typ host generation 0
jssip-0.3.0.js:3369
JsSIP | RTC SESSION | ICE candidate received: a=candidate:2999745851 2 udp 2113937151
192.168.56.1 65224 typ host generation 0
jssip-0.3.0.js:3369
JsSIP | RTC SESSION | ICE candidate received: a=candidate:1757736706 1 tcp 1509957375
10.10.10.63 0 typ host generation 0
jssip-0.3.0.js:3369
JsSIP | RTC SESSION | ICE candidate received: a=candidate:1757736706 2 tcp 1509957375
10.10.10.63 0 typ host generation 0
jssip-0.3.0.js:3369
JsSIP | RTC SESSION | ICE candidate received: a=candidate:4233069003 1 tcp 1509957375
192.168.56.1 0 typ host generation 0
jssip-0.3.0.js:3369
JsSIP | RTC SESSION | ICE candidate received: a=candidate:4233069003 2 tcp 1509957375
192.168.56.1 0 typ host generation 0
jssip-0.3.0.js:3369
JsSIP | TRANSPORT | sending WebSocket message:
INVITE sip:9822@10.1.1.101 SIP/2.0
Via: SIP/2.0/WSS flogvr403sb2.invalid;branch=z9hG4bK9149581
Max-Forwards: 69
To: <sip:9822@10.1.1.101>
From: <sip:webrtc@10.10.10.48>;tag=6tmeble9ov
Call-ID: 43oclsi0sva6n347bk5c
CSeq: 7435 INVITE
Contact: <sip:ce5egl03@flogvr403sb2.invalid;transport=ws;ob>
Allow: ACK,CANCEL,BYE,OPTIONS,INVITE
Content-Type: application/sdp
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.0
Content-Length: 1744
v=0
o=- 3746191339358890844 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS TNUolHZksseiQbV1o2j8kmZGxOkOjYsZYXh8
m=audio 65223 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 10.10.10.63
a=rtcp:65223 IN IP4 10.10.10.63
a=candidate:642192370 1 udp 2113937151 10.10.10.63 65223 typ host generation 0
a=candidate:642192370 2 udp 2113937151 10.10.10.63 65223 typ host generation 0
a=candidate:2999745851 1 udp 2113937151 192.168.56.1 65224 typ host generation 0
a=candidate:2999745851 2 udp 2113937151 192.168.56.1 65224 typ host generation 0
a=candidate:1757736706 1 tcp 1509957375 10.10.10.63 0 typ host generation 0
a=candidate:1757736706 2 tcp 1509957375 10.10.10.63 0 typ host generation 0
a=candidate:4233069003 1 tcp 1509957375 192.168.56.1 0 typ host generation 0
a=candidate:4233069003 2 tcp 1509957375 192.168.56.1 0 typ host generation 0
a=ice-ufrag:Dgp8HIJdmr1lFPCQ
a=ice-pwd:2yYxerrscdbTQhr0vbCTiju9
a=ice-options:google-ice
a=fingerprint:sha-256
C8:E9:57:CB:85:63:F7:C5:FC:15:3D:8B:A8:10:94:F4:C9:BB:86:48:E3:EE:A0:5E:FA:42:14:55:6F:68:3F:B7
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
a=ssrc:3445528109 cname:QbLF+sVLqHbEqUxY
a=ssrc:3445528109 msid:TNUolHZksseiQbV1o2j8kmZGxOkOjYsZYXh8
768e2e47-bc86-473d-bc2c-6e2340ace772
a=ssrc:3445528109 mslabel:TNUolHZksseiQbV1o2j8kmZGxOkOjYsZYXh8
a=ssrc:3445528109 label:768e2e47-bc86-473d-bc2c-6e2340ace772
jssip-0.3.0.js:519
JsSIP | TRANSPORT | received WebSocket text message:
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/WSS
flogvr403sb2.invalid;branch=z9hG4bK9149581;rport=56527;received=10.10.10.63
To: <sip:9822@10.1.1.101>
From: <sip:webrtc@10.10.10.48>;tag=6tmeble9ov
Call-ID: 43oclsi0sva6n347bk5c
CSeq: 7435 INVITE
Server: DXI WebRTC
Content-Length: 0
Warning: 392 10.10.10.48:6443 "Noisy feedback tells: pid=23455
req_src_ip=10.10.10.63 req_src_port=56527 in_uri=sip:9822@10.1.1.101
out_uri=sip:9822@10.10.10.111:5443 via_cnt==1"
jssip-0.3.0.js:670
JsSIP | TRANSPORT | received WebSocket text message:
SIP/2.0 200 OK
Via: SIP/2.0/WSS
flogvr403sb2.invalid;rport=56527;received=10.10.10.63;branch=z9hG4bK9149581
From: <sip:webrtc@10.10.10.48>;tag=6tmeble9ov
To: <sip:9822@10.1.1.101>;tag=as06b3db08
Call-ID: 43oclsi0sva6n347bk5c
CSeq: 7435 INVITE
Server: Easycall
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH,
MESSAGE
Supported: replaces, timer
Contact: <sip:9822@10.10.10.111:5443;transport=TLS>
Content-Type: application/sdp
Content-Length: 801
v=0
o=root 431209641 431209641 IN IP4 10.10.10.111
s=Asterisk PBX 12.2.0-rc1
c=IN IP4 10.10.10.111
t=0 0
m=audio 30490 UDP/TLS/RTP/SAVPF 0 126
a=rtpmap:0 PCMU/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=maxptime:150
a=ice-ufrag:5236e84b43b5d10c117e8ead0a340138
a=ice-pwd:51005fde4e6e9d3f1879fbbc15e0f092
a=candidate:Ha1f026f 1 UDP 2130706431 10.10.10.111 30490 typ host
a=candidate:S5bec7504 1 UDP 1694498815 91.236.117.4 30490 typ srflx
a=candidate:Ha1f026f 2 UDP 2130706430 10.10.10.111 30491 typ host
a=candidate:S5bec7504 2 UDP 1694498814 91.236.117.4 30492 typ srflx
a=connection:new
a=setup:active
a=fingerprint:SHA-256
13:BB:CF:88:C4:75:9B:F0:DA:36:0A:6D:5D:37:C9:26:6B:3C:82:3E:F6:92:AE:A7:AE:CF:FF:78:F5:86:D9:E8
a=sendrecv
jssip-0.3.0.js:670
Failed to parse SessionDescription. Failed to parse audio codecs correctly.
jssip-0.3.0.js:4512
JsSIP | DIALOG | new UAC dialog created with status CONFIRMED jssip-0.3.0.js:2523
JsSIP | TRANSPORT | sending WebSocket message:
ACK sip:9822@10.10.10.111:5443;transport=tls SIP/2.0
Via: SIP/2.0/WSS flogvr403sb2.invalid;branch=z9hG4bK431640
Max-Forwards: 69
To: <sip:9822@10.1.1.101>;tag=as06b3db08
From: <sip:webrtc@10.10.10.48>;tag=6tmeble9ov
Call-ID: 43oclsi0sva6n347bk5c
CSeq: 7435 ACK
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.0
Content-Length: 0
jssip-0.3.0.js:519
JsSIP | TRANSPORT | sending WebSocket message:
BYE sip:9822@10.10.10.111:5443;transport=tls SIP/2.0
Via: SIP/2.0/WSS flogvr403sb2.invalid;branch=z9hG4bK7689766
Max-Forwards: 69
To: <sip:9822@10.1.1.101>;tag=as06b3db08
From: <sip:webrtc@10.10.10.48>;tag=6tmeble9ov
Call-ID: 43oclsi0sva6n347bk5c
CSeq: 7436 BYE
Reason: SIP ;cause=488; text="Not Acceptable Here"
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.0
Content-Length: 0
jssip-0.3.0.js:519
JsSIP | RTC SESSION | closing INVITE session 43oclsi0sva6n347bk5c6tmeble9ov
jssip-0.3.0.js:4193
JsSIP | RTC SESSION | closing PeerConnection jssip-0.3.0.js:3392
JsSIP | DIALOG | dialog 43oclsi0sva6n347bk5c6tmeble9ovas06b3db08 deleted
jssip-0.3.0.js:2543
JsSIP | EVENT EMITTER | emitting event failed jssip-0.3.0.js:187
JsSIP | TRANSPORT | received WebSocket text message:
SIP/2.0 200 OK
Via: SIP/2.0/WSS
flogvr403sb2.invalid;rport=56527;received=10.10.10.63;branch=z9hG4bK7689766
From: <sip:webrtc@10.10.10.48>;tag=6tmeble9ov
To: <sip:9822@10.1.1.101>;tag=as06b3db08
Call-ID: 43oclsi0sva6n347bk5c
CSeq: 7436 BYE
Server: Easycall
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH,
MESSAGE
Supported: replaces, timer
Content-Length: 0
_______________________________________________
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sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
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9:30 a.m.
upps ... sorry ... *pass th**rough* and not path through :-[
2013-08-23 15:49 +0000 [r397524-397527] Matthew Jordan <mjordan(a)digium.com>
* CHANGES: Update CHANGES file to reflect pass through support for
Opus/VP8
Source ->
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.…
Regards
Rainer
Am 04.04.2014 08:18, schrieb Rainer Piper:
Hallo,
my guess is the audio codec opus
asterisk can NOT do transcoding from opus to pcmu.
The opus codec in asterisk is (just) a path through codec.
your trace right at the end:
!!! Failed to parse SessionDescription. Failed to parse audio codecs correctly !!!
Regards
Rainer
Am 03.04.2014 18:11, schrieb Richard Fuchs:
--
*Rainer Piper*
NOC - +49 (0)228 97167161 - sip.soho-piper.de
NOC - +49 (0)2247 9064188 - sip.tele33.de - sip.tefonix.de - D293
My guess would be that it's due to a
discrepancy between WebRTC and RFC
5764. WebRTC uses a protocol string of "RTP/SAVPF", while the RFC says
that "UDP/TLS/RTP/SAVPF" shall be used. You can try an SDP rewrite
operation to substitute one for the other. Or teach your non-RTC client
to use a different protocol string.
cheers
On 04/03/14 10:22, jaflong jaflong wrote:
--
*Rainer Piper*
NOC - +49 (0)228 97167161 <callto:004922897167161> - sip.soho-piper.de
NOC - +49 (0)2247 9064188 <callto:004922479064188> - sip.tele33.de -
sip.tefonix.de - D293
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users Hi List,
Can anyone help me understand why this is getting rejected
Please note the specific message further dow the log.
"Failed to parse SessionDescription. Failed to parse audio codecs correctly"
This is on Chrome.
On Firefox There is a further message in the console
"Could not negotiate answer SDP; cause = ERR | SDP Parsing Error: Warning:
Transport protocol type unsupported (UDP/TLS/RTP/SAVPF). | SDP Parsing Error: Invalid
port format (17296) specified for transport protocol (Unsupported), parse failed."
JsSIP | RTC SESSION | got local media stream jssip-0.3.0.js:3414
JsSIP | RTC SESSION | ICE candidate received: a=candidate:642192370 1 udp 2113937151
10.10.10.63 65223 typ host generation 0
jssip-0.3.0.js:3369
JsSIP | RTC SESSION | ICE candidate received: a=candidate:642192370 2 udp 2113937151
10.10.10.63 65223 typ host generation 0
jssip-0.3.0.js:3369
JsSIP | RTC SESSION | ICE candidate received: a=candidate:2999745851 1 udp 2113937151
192.168.56.1 65224 typ host generation 0
jssip-0.3.0.js:3369
JsSIP | RTC SESSION | ICE candidate received: a=candidate:2999745851 2 udp 2113937151
192.168.56.1 65224 typ host generation 0
jssip-0.3.0.js:3369
JsSIP | RTC SESSION | ICE candidate received: a=candidate:1757736706 1 tcp 1509957375
10.10.10.63 0 typ host generation 0
jssip-0.3.0.js:3369
JsSIP | RTC SESSION | ICE candidate received: a=candidate:1757736706 2 tcp 1509957375
10.10.10.63 0 typ host generation 0
jssip-0.3.0.js:3369
JsSIP | RTC SESSION | ICE candidate received: a=candidate:4233069003 1 tcp 1509957375
192.168.56.1 0 typ host generation 0
jssip-0.3.0.js:3369
JsSIP | RTC SESSION | ICE candidate received: a=candidate:4233069003 2 tcp 1509957375
192.168.56.1 0 typ host generation 0
jssip-0.3.0.js:3369
JsSIP | TRANSPORT | sending WebSocket message:
INVITEsip:9822@10.1.1.101 SIP/2.0
Via: SIP/2.0/WSS flogvr403sb2.invalid;branch=z9hG4bK9149581
Max-Forwards: 69
To:<sip:9822@10.1.1.101>
From:<sip:webrtc@10.10.10.48>;tag=6tmeble9ov
Call-ID: 43oclsi0sva6n347bk5c
CSeq: 7435 INVITE
Contact:<sip:ce5egl03@flogvr403sb2.invalid;transport=ws;ob>
Allow: ACK,CANCEL,BYE,OPTIONS,INVITE
Content-Type: application/sdp
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.0
Content-Length: 1744
v=0
o=- 3746191339358890844 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS TNUolHZksseiQbV1o2j8kmZGxOkOjYsZYXh8
m=audio 65223 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 10.10.10.63
a=rtcp:65223 IN IP4 10.10.10.63
a=candidate:642192370 1 udp 2113937151 10.10.10.63 65223 typ host generation 0
a=candidate:642192370 2 udp 2113937151 10.10.10.63 65223 typ host generation 0
a=candidate:2999745851 1 udp 2113937151 192.168.56.1 65224 typ host generation 0
a=candidate:2999745851 2 udp 2113937151 192.168.56.1 65224 typ host generation 0
a=candidate:1757736706 1 tcp 1509957375 10.10.10.63 0 typ host generation 0
a=candidate:1757736706 2 tcp 1509957375 10.10.10.63 0 typ host generation 0
a=candidate:4233069003 1 tcp 1509957375 192.168.56.1 0 typ host generation 0
a=candidate:4233069003 2 tcp 1509957375 192.168.56.1 0 typ host generation 0
a=ice-ufrag:Dgp8HIJdmr1lFPCQ
a=ice-pwd:2yYxerrscdbTQhr0vbCTiju9
a=ice-options:google-ice
a=fingerprint:sha-256
C8:E9:57:CB:85:63:F7:C5:FC:15:3D:8B:A8:10:94:F4:C9:BB:86:48:E3:EE:A0:5E:FA:42:14:55:6F:68:3F:B7
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
a=ssrc:3445528109 cname:QbLF+sVLqHbEqUxY
a=ssrc:3445528109 msid:TNUolHZksseiQbV1o2j8kmZGxOkOjYsZYXh8
768e2e47-bc86-473d-bc2c-6e2340ace772
a=ssrc:3445528109 mslabel:TNUolHZksseiQbV1o2j8kmZGxOkOjYsZYXh8
a=ssrc:3445528109 label:768e2e47-bc86-473d-bc2c-6e2340ace772
jssip-0.3.0.js:519
JsSIP | TRANSPORT | received WebSocket text message:
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/WSS
flogvr403sb2.invalid;branch=z9hG4bK9149581;rport=56527;received=10.10.10.63
To:<sip:9822@10.1.1.101>
From:<sip:webrtc@10.10.10.48>;tag=6tmeble9ov
Call-ID: 43oclsi0sva6n347bk5c
CSeq: 7435 INVITE
Server: DXI WebRTC
Content-Length: 0
Warning: 392 10.10.10.48:6443 "Noisy feedback tells: pid=23455
req_src_ip=10.10.10.63 req_src_port=56527 in_uri=sip:9822@10.1.1.101
out_uri=sip:9822@10.10.10.111:5443 via_cnt==1"
jssip-0.3.0.js:670
JsSIP | TRANSPORT | received WebSocket text message:
SIP/2.0 200 OK
Via: SIP/2.0/WSS
flogvr403sb2.invalid;rport=56527;received=10.10.10.63;branch=z9hG4bK9149581
From:<sip:webrtc@10.10.10.48>;tag=6tmeble9ov
To:<sip:9822@10.1.1.101>;tag=as06b3db08
Call-ID: 43oclsi0sva6n347bk5c
CSeq: 7435 INVITE
Server: Easycall
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH,
MESSAGE
Supported: replaces, timer
Contact:<sip:9822@10.10.10.111:5443;transport=TLS>
Content-Type: application/sdp
Content-Length: 801
v=0
o=root 431209641 431209641 IN IP4 10.10.10.111
s=Asterisk PBX 12.2.0-rc1
c=IN IP4 10.10.10.111
t=0 0
m=audio 30490 UDP/TLS/RTP/SAVPF 0 126
a=rtpmap:0 PCMU/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=maxptime:150
a=ice-ufrag:5236e84b43b5d10c117e8ead0a340138
a=ice-pwd:51005fde4e6e9d3f1879fbbc15e0f092
a=candidate:Ha1f026f 1 UDP 2130706431 10.10.10.111 30490 typ host
a=candidate:S5bec7504 1 UDP 1694498815 91.236.117.4 30490 typ srflx
a=candidate:Ha1f026f 2 UDP 2130706430 10.10.10.111 30491 typ host
a=candidate:S5bec7504 2 UDP 1694498814 91.236.117.4 30492 typ srflx
a=connection:new
a=setup:active
a=fingerprint:SHA-256
13:BB:CF:88:C4:75:9B:F0:DA:36:0A:6D:5D:37:C9:26:6B:3C:82:3E:F6:92:AE:A7:AE:CF:FF:78:F5:86:D9:E8
a=sendrecv
jssip-0.3.0.js:670
Failed to parse SessionDescription. Failed to parse audio codecs correctly.
jssip-0.3.0.js:4512
JsSIP | DIALOG | new UAC dialog created with status CONFIRMED jssip-0.3.0.js:2523
JsSIP | TRANSPORT | sending WebSocket message:
ACKsip:9822@10.10.10.111:5443;transport=tls SIP/2.0
Via: SIP/2.0/WSS flogvr403sb2.invalid;branch=z9hG4bK431640
Max-Forwards: 69
To:<sip:9822@10.1.1.101>;tag=as06b3db08
From:<sip:webrtc@10.10.10.48>;tag=6tmeble9ov
Call-ID: 43oclsi0sva6n347bk5c
CSeq: 7435 ACK
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.0
Content-Length: 0
jssip-0.3.0.js:519
JsSIP | TRANSPORT | sending WebSocket message:
BYEsip:9822@10.10.10.111:5443;transport=tls SIP/2.0
Via: SIP/2.0/WSS flogvr403sb2.invalid;branch=z9hG4bK7689766
Max-Forwards: 69
To:<sip:9822@10.1.1.101>;tag=as06b3db08
From:<sip:webrtc@10.10.10.48>;tag=6tmeble9ov
Call-ID: 43oclsi0sva6n347bk5c
CSeq: 7436 BYE
Reason: SIP ;cause=488; text="Not Acceptable Here"
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.0
Content-Length: 0
jssip-0.3.0.js:519
JsSIP | RTC SESSION | closing INVITE session 43oclsi0sva6n347bk5c6tmeble9ov
jssip-0.3.0.js:4193
JsSIP | RTC SESSION | closing PeerConnection jssip-0.3.0.js:3392
JsSIP | DIALOG | dialog 43oclsi0sva6n347bk5c6tmeble9ovas06b3db08 deleted
jssip-0.3.0.js:2543
JsSIP | EVENT EMITTER | emitting event failed jssip-0.3.0.js:187
JsSIP | TRANSPORT | received WebSocket text message:
SIP/2.0 200 OK
Via: SIP/2.0/WSS
flogvr403sb2.invalid;rport=56527;received=10.10.10.63;branch=z9hG4bK7689766
From:<sip:webrtc@10.10.10.48>;tag=6tmeble9ov
To:<sip:9822@10.1.1.101>;tag=as06b3db08
Call-ID: 43oclsi0sva6n347bk5c
CSeq: 7436 BYE
Server: Easycall
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH,
MESSAGE
Supported: replaces, timer
Content-Length: 0
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
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10:29 a.m.
El Fri, 04 Apr 2014 08:18:22 +0200
Rainer Piper <rainer.piper(a)soho-piper.de> escribió:
Hallo,
my guess is the audio codec opus
asterisk can NOT do transcoding from opus to pcmu.
The opus codec in asterisk is (just) a path through codec.
your trace right at the end:
!!! Failed to parse SessionDescription. Failed to parse audio codecs
correctly !!!
Just in case you don't know the patch:
https://github.com/meetecho/asterisk-opus
cheers,
Jon
10:31 a.m.
Hi,
Is it possible to use pcmu start to end, so I send pcmu instead of opus from the browser?
Regards
04.04.2014, 12:29, "Jon Bonilla (Manwe)" <manwe(a)aholab.ehu.es>es>:
El Fri, 04 Apr 2014 08:18:22 +0200
Rainer Piper <rainer.piper(a)soho-piper.de> escribió:
Hallo,
my guess is the audio codec opus
asterisk can NOT do transcoding from opus to pcmu.
The opus codec in asterisk is (just) a path through codec.
your trace right at the end:
!!! Failed to parse SessionDescription. Failed to parse audio codecs
correctly !!!
Just in case you don't know the patch:
https://github.com/meetecho/asterisk-opus
cheers,
Jon
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10:54 a.m.
you can try to turn of the opus codec support in the browser.
At firefox ... open about:config and search for media.opus.enabled and
set it to false.
At Chrome ... open about:flags and search for opus.
Regards
Rainer
Am 04.04.2014 10:31, schrieb jaflong jaflong:
Hi,
Is it possible to use pcmu start to end, so I send pcmu instead of opus from the
browser?
Regards
04.04.2014, 12:29, "Jon Bonilla (Manwe)" <manwe(a)aholab.ehu.es>es>:
--
*Rainer Piper*
NOC - +49 (0)228 97167161 - sip.soho-piper.de
NOC - +49 (0)2247 9064188 - sip.tele33.de - sip.tefonix.de - D293
El Fri, 04 Apr 2014 08:18:22 +0200
Rainer Piper <rainer.piper(a)soho-piper.de> escribió:
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SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
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http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users Hallo,
my guess is the audio codec opus
asterisk can NOT do transcoding from opus to pcmu.
The opus codec in asterisk is (just) a path through codec.
your trace right at the end:
!!! Failed to parse SessionDescription. Failed to parse audio codecs
correctly !!!
Just in case you don't know the patch:
https://github.com/meetecho/asterisk-opus
cheers,
Jon
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6 comments
4 participants
participants (4)
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jaflong jaflong -
Jon Bonilla (Manwe) -
Rainer Piper -
Richard Fuchs