[jitsi-dev] iOS etc


#1

Hello,

I fixed whitespace and changed end of line from \n to \r\n iike the java code. no change though.

i found that i was not putting these in for video candidates
  a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
  a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time

i have no lines like this translated from jingle but the sdp from appspot server sdp does.
  c=IN IP4 217.78.97.15
  a=rtcp:57026 IN IP4 217.78.97.15

i have ice-frag and ice-pwd before candidates and appspot example has it after so i switched order.

NOW i get this:
Error(webrtcsdp.cc:357): Failed to parse: "". Reason: Failed to parse audio codecs correctly.

IF there is anything else you need to see to help get to the bottom of this, please let me know.

new sdp looks like this
SDP SO FAR IS
v=0

o=- 9135848008193788882 2 IN IP4 127.0.0.1

s=-

t=0 0

a=group:BUNDLE audio video

m=audio RTP/SAVPF 111 103 104 0 8 106 105 13 126

a=candidate:1 1 udp 2130706431 2002:902:37b8:0:0:0:902:37b8 5439 typ host

a=candidate:2 1 udp 2130706431 9.2.55.184 5439 typ host

a=candidate:3 1 udp 2113937151 fe80:0:0:0:e0e1:bee7:27dd:a0a1 5439 typ host

a=candidate:1 2 udp 2130706430 2002:902:37b8:0:0:0:902:37b8 5440 typ host

a=candidate:2 2 udp 2130706430 9.2.55.184 5440 typ host

a=candidate:3 2 udp 2113937150 fe80:0:0:0:e0e1:bee7:27dd:a0a1 5440 typ host

a=ice-ufrag:d2go5

a=ice-pwd:69unsj2efofr85thvjagj85cae

a=mid:audio

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=ssrc:1027749449 cname:20vbtzsaI2W/gqQf

a=ssrc:1027749449 msid:Ji5HVTyO3vAprIERrnfQh29IcuqBuKvgexr4 7075601e-0ceb-48e3-b13d-ea8a9324f160

a=ssrc:1027749449 mslabel:Ji5HVTyO3vAprIERrnfQh29IcuqBuKvgexr4

a=ssrc:1027749449 label:7075601e-0ceb-48e3-b13d-ea8a9324f160

a=ssrc:3735928559 cname:mixed

a=ssrc:3735928559 label:mixedlabelv0

a=ssrc:3735928559 msid:mixedmslabel mixedlabelv0

a=ssrc:3735928559 mslabel:mixedmslabel

a=fingerprint:sha-1 E7:38:F7:2F:12:07:C5:10:CA:3C:21:F9:47:D2:A0:C5:09:51:30:C3

m=video RTP/SAVPF 100 116 117

a=candidate:1 1 udp 2130706431 2002:902:37b8:0:0:0:902:37b8 5443 typ host

a=candidate:2 1 udp 2130706431 9.2.55.184 5443 typ host

a=candidate:3 1 udp 2113937151 fe80:0:0:0:e0e1:bee7:27dd:a0a1 5443 typ host

a=candidate:1 2 udp 2130706430 2002:902:37b8:0:0:0:902:37b8 5444 typ host

a=candidate:2 2 udp 2130706430 9.2.55.184 5444 typ host

a=candidate:3 2 udp 2113937150 fe80:0:0:0:e0e1:bee7:27dd:a0a1 5444 typ host

a=ice-ufrag:3ls50

a=ice-pwd:5sqtfk8bflm9id28qnakcnd0km

a=mid:video

a=extmap:2 urn:ietf:params:rtp-hdrext:toffset

a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time

a=sendrecv

a=rtcp-mux

a=rtpmap:100 VP8/90000

a=rtpmap:116 red/90000

a=rtpmap:117 ulpfec/90000

a=ssrc:3333184907 cname:2t951WLKfY9O8W3h

a=ssrc:3333184907 msid:ABDQ3PByI8HAsSw1db2nRU7rlpzEzCB22Ohe 45ed8da3-32fd-4580-86a5-34233abcb46c

a=ssrc:3333184907 mslabel:ABDQ3PByI8HAsSw1db2nRU7rlpzEzCB22Ohe

a=ssrc:3333184907 label:45ed8da3-32fd-4580-86a5-34233abcb46c

a=ssrc:3735928559 cname:mixed

a=ssrc:3735928559 label:mixedlabelv0

a=ssrc:3735928559 msid:mixedmslabel mixedlabelv0

a=ssrc:3735928559 mslabel:mixedmslabel

a=fingerprint:sha-1 5A:B7:C9:86:40:3A:96:38:D2:41:11:43:76:1C:62:5D:94:F6:20:37

2014-11-06 14:49:59.227 AppRTCDemo[3403:1019125] SEND: <presence type="unavailable"><x xmlns="vcard-temp:x:update"><photo/></x></presence>
2014-11-06 14:49:59.252 AppRTCDemo[3403:1019125] ---------- xmppStreamDidDisconnect: ----------
2014-11-06 14:49:59.258 AppRTCDemo[3403:1019125] UITableView has 2 rows
2014-11-06 14:50:00.139 AppRTCDemo[3403:1019125] SEQ2-Sending CONNECT to room # 98303690
2014-11-06 14:50:00.140 AppRTCDemo[3403:1019125] *** HERE in acceptConnection
Warning(webrtcvoiceengine.cc:360): SetTraceCallback() failed, err=0
WebRtc VoiceEngine codecs:
ISAC/16000/1 (103)
PCMU/8000/1 (0)
PCMA/8000/1 (8)
Warning(webrtcvoiceengine.cc:462): Unexpected codec: PCMU/8000/2 (110)
Warning(webrtcvoiceengine.cc:462): Unexpected codec: PCMA/8000/2 (118)
ILBC/8000/1 (102)
G722/16000/1 (9)
Warning(webrtcvoiceengine.cc:462): Unexpected codec: G722/16000/2 (119)
opus/48000/2 (111)
CN/8000/1 (13)
CN/16000/1 (105)
CN/32000/1 (106)
telephone-event/8000/1 (126)
red/8000/1 (127)
WebRtcVideoEngine::WebRtcVideoEngine
Warning(webrtcvideoengine.cc:838): SetTraceCallback(0x15c51e5c) failed, err=0
WebRtcVoiceEngine::Init
WebRtc VoiceEngine Version:
VoiceEngine 4.1.0
Build: svn:Unavailable(issue687) Oct 29 2013 12:59:30 ?
Applying audio options: AudioOptions {aec: false, agc: false, ns: true, hf: true, swap: false, typing: false, conference: false, agc_delta: 0, experimental_agc: false, experimental_aec: false, aec_dump: false, }
Warning(webrtcvoiceengine.cc:811): SetTypingDetectionStatus(0) failed, err=8003
Adjusting AGC level from default -3dB to -3dB
WebRtc VoiceEngine codecs:
opus/48000/2 (111)
ISAC/16000/1 (103)
G722/16000/1 (9)
ILBC/8000/1 (102)
PCMU/8000/1 (0)
PCMA/8000/1 (8)
CN/32000/1 (106)
CN/16000/1 (105)
CN/8000/1 (13)
red/8000/1 (127)
telephone-event/8000/1 (126)
WebRtcVoiceEngine::Init Done!
WebRtcVideoEngine::Init
WebRtcVideoEngine::InitVideoEngine
WebRtc VideoEngine Version:
VideoEngine 3.45.0
Build: svn:Unavailable(issue687) Oct 29 2013 13:00:07 ?
VideoEngine Init done
Applying audio options: AudioOptions {aec: false, agc: false, ns: true, hf: true, swap: false, typing: false, conference: false, agc_delta: 0, experimental_agc: false, experimental_aec: false, aec_dump: false, }
Warning(webrtcvoiceengine.cc:811): SetTypingDetectionStatus(0) failed, err=8003
Adjusting AGC level from default -3dB to -3dB
Allowing SCTP data engine.
Generating identity.
2014-11-06 14:50:01.616 AppRTCDemo[3403:1019125] PC - setRemoteDescription.
Ignored line: a=sendrecv
Ignored line: a=rtcp-mux
Ignored line: a=rtpmap:111 opus/48000/2
Ignored line: a=rtpmap:103 ISAC/16000
Ignored line: a=rtpmap:104 ISAC/32000
Ignored line: a=rtpmap:0 PCMU/8000
Ignored line: a=rtpmap:8 PCMA/8000
Ignored line: a=rtpmap:106 CN/32000
Ignored line: a=rtpmap:105 CN/16000
Ignored line: a=rtpmap:13 CN/8000
Ignored line: a=rtpmap:126 telephone-event/8000
Ignored line: a=ssrc:1027749449 cname:20vbtzsaI2W/gqQf
Ignored line: a=ssrc:1027749449 msid:Ji5HVTyO3vAprIERrnfQh29IcuqBuKvgexr4 7075601e-0ceb-48e3-b13d-ea8a9324f160
Ignored line: a=ssrc:1027749449 mslabel:Ji5HVTyO3vAprIERrnfQh29IcuqBuKvgexr4
Ignored line: a=ssrc:1027749449 label:7075601e-0ceb-48e3-b13d-ea8a9324f160
Ignored line: a=ssrc:3735928559 cname:mixed
Ignored line: a=ssrc:3735928559 label:mixedlabelv0
Ignored line: a=ssrc:3735928559 msid:mixedmslabel mixedlabelv0
Ignored line: a=ssrc:3735928559 mslabel:mixedmslabel
Error(webrtcsdp.cc:357): Failed to parse: "". Reason: Failed to parse audio codecs correctly.
2014-11-06 14:50:03.852 AppRTCDemo[3403:1019125] SEQ3-Connect button pressed ...
2014-11-06 14:50:03.852 AppRTCDemo[3403:1019125] SEQ4-Setup AppRTC video view
2014-11-06 14:50:03.852 AppRTCDemo[3403:1019276] SDP onFailure.
(lldb)


#2

Hello,

I fixed whitespace and changed end of line from \n to \r\n iike the java
code. no change though.

i found that i was not putting these in for video candidates
a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time

i have no lines like this translated from jingle but the sdp from
appspot server sdp does.
c=IN IP4 217.78.97.15
a=rtcp:57026 IN IP4 217.78.97.15

See ttps://github.com/jitsi/jitsi-meet/blob/master/libs/strophe/strophe.jingle.sdp.js#L579
You can put the same dummy values here.

i have ice-frag and ice-pwd before candidates and appspot example has it
after so i switched order.

NOW i get this:
*Error(webrtcsdp.cc <http://webrtcsdp.cc>:357): Failed to parse: "".
Reason: Failed to parse audio codecs correctly.*

That might be caused by the malformed m-line emil pointed out.

···

Am 06.11.2014 um 11:54 schrieb Peter Mycue:


#3

You are missing tue port number in the audio and video m=lined. I think
Chrome sets this to 1 currently but it'supposed to be 9. In other words:

m=audio 1 RTP/SAVPF 111 103 104 0 8 106 105 13 126

--sent from my mobile

···

On 06 Nov 2014 8:56 PM, "Peter Mycue" <pmycue@gmail.com> wrote:

Hello,

I fixed whitespace and changed end of line from \n to \r\n iike the java
code. no change though.

i found that i was not putting these in for video candidates
a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time

i have no lines like this translated from jingle but the sdp from appspot
server sdp does.
c=IN IP4 217.78.97.15
a=rtcp:57026 IN IP4 217.78.97.15

i have ice-frag and ice-pwd before candidates and appspot example has it
after so i switched order.

NOW i get this:
*Error(webrtcsdp.cc <http://webrtcsdp.cc>:357): Failed to parse: "".
Reason: Failed to parse audio codecs correctly.*

*IF there is anything else you need to see to help get to the bottom of
this, please let me know.*

new sdp looks like this
*SDP SO FAR IS*
*v=0*

*o=- 9135848008193788882 2 IN IP4 127.0.0.1*

*s=-*

*t=0 0*

*a=group:BUNDLE audio video*

*m=audio RTP/SAVPF 111 103 104 0 8 106 105 13 126*

*a=candidate:1 1 udp 2130706431 2002:902:37b8:0:0:0:902:37b8 5439 typ host*

*a=candidate:2 1 udp 2130706431 9.2.55.184 5439 typ host*

*a=candidate:3 1 udp 2113937151 fe80:0:0:0:e0e1:bee7:27dd:a0a1 5439 typ
host*

*a=candidate:1 2 udp 2130706430 2002:902:37b8:0:0:0:902:37b8 5440 typ host*

*a=candidate:2 2 udp 2130706430 9.2.55.184 5440 typ host*

*a=candidate:3 2 udp 2113937150 fe80:0:0:0:e0e1:bee7:27dd:a0a1 5440 typ
host*

*a=ice-ufrag:d2go5*

*a=ice-pwd:69unsj2efofr85thvjagj85cae*

*a=mid:audio*

*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=ssrc:1027749449 cname:20vbtzsaI2W/gqQf*

*a=ssrc:1027749449 msid:Ji5HVTyO3vAprIERrnfQh29IcuqBuKvgexr4
7075601e-0ceb-48e3-b13d-ea8a9324f160*

*a=ssrc:1027749449 mslabel:Ji5HVTyO3vAprIERrnfQh29IcuqBuKvgexr4*

*a=ssrc:1027749449 label:7075601e-0ceb-48e3-b13d-ea8a9324f160*

*a=ssrc:3735928559 cname:mixed*

*a=ssrc:3735928559 label:mixedlabelv0*

*a=ssrc:3735928559 msid:mixedmslabel mixedlabelv0*

*a=ssrc:3735928559 mslabel:mixedmslabel*

*a=fingerprint:sha-1
E7:38:F7:2F:12:07:C5:10:CA:3C:21:F9:47:D2:A0:C5:09:51:30:C3*

*m=video RTP/SAVPF 100 116 117*

*a=candidate:1 1 udp 2130706431 2002:902:37b8:0:0:0:902:37b8 5443 typ host*

*a=candidate:2 1 udp 2130706431 9.2.55.184 5443 typ host*

*a=candidate:3 1 udp 2113937151 fe80:0:0:0:e0e1:bee7:27dd:a0a1 5443 typ
host*

*a=candidate:1 2 udp 2130706430 2002:902:37b8:0:0:0:902:37b8 5444 typ host*

*a=candidate:2 2 udp 2130706430 9.2.55.184 5444 typ host*

*a=candidate:3 2 udp 2113937150 fe80:0:0:0:e0e1:bee7:27dd:a0a1 5444 typ
host*

*a=ice-ufrag:3ls50*

*a=ice-pwd:5sqtfk8bflm9id28qnakcnd0km*

*a=mid:video*

*a=extmap:2 urn:ietf:params:rtp-hdrext:toffset*

*a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
<http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time>*

*a=sendrecv*

*a=rtcp-mux*

*a=rtpmap:100 VP8/90000*

*a=rtpmap:116 red/90000*

*a=rtpmap:117 ulpfec/90000*

*a=ssrc:3333184907 cname:2t951WLKfY9O8W3h*

*a=ssrc:3333184907 msid:ABDQ3PByI8HAsSw1db2nRU7rlpzEzCB22Ohe
45ed8da3-32fd-4580-86a5-34233abcb46c*

*a=ssrc:3333184907 mslabel:ABDQ3PByI8HAsSw1db2nRU7rlpzEzCB22Ohe*

*a=ssrc:3333184907 label:45ed8da3-32fd-4580-86a5-34233abcb46c*

*a=ssrc:3735928559 cname:mixed*

*a=ssrc:3735928559 label:mixedlabelv0*

*a=ssrc:3735928559 msid:mixedmslabel mixedlabelv0*

*a=ssrc:3735928559 mslabel:mixedmslabel*

*a=fingerprint:sha-1
5A:B7:C9:86:40:3A:96:38:D2:41:11:43:76:1C:62:5D:94:F6:20:37*

*2014-11-06 14:49:59.227 AppRTCDemo[3403:1019125] SEND: <presence
type="unavailable"><x xmlns="vcard-temp:x:update"><photo/></x></presence>*
*2014-11-06 14:49:59.252 AppRTCDemo[3403:1019125] ----------
xmppStreamDidDisconnect: ----------*
*2014-11-06 14:49:59.258 AppRTCDemo[3403:1019125] UITableView has 2 rows*
*2014-11-06 14:50:00.139 AppRTCDemo[3403:1019125] SEQ2-Sending CONNECT to
room # 98303690*
*2014-11-06 14:50:00.140 AppRTCDemo[3403:1019125] *** HERE in
acceptConnection*
*Warning(webrtcvoiceengine.cc <http://webrtcvoiceengine.cc>:360):
SetTraceCallback() failed, err=0*
*WebRtc VoiceEngine codecs:*
*ISAC/16000/1 (103)*
*PCMU/8000/1 (0)*
*PCMA/8000/1 (8)*
*Warning(webrtcvoiceengine.cc <http://webrtcvoiceengine.cc>:462):
Unexpected codec: PCMU/8000/2 (110)*
*Warning(webrtcvoiceengine.cc <http://webrtcvoiceengine.cc>:462):
Unexpected codec: PCMA/8000/2 (118)*
*ILBC/8000/1 (102)*
*G722/16000/1 (9)*
*Warning(webrtcvoiceengine.cc <http://webrtcvoiceengine.cc>:462):
Unexpected codec: G722/16000/2 (119)*
*opus/48000/2 (111)*
*CN/8000/1 (13)*
*CN/16000/1 (105)*
*CN/32000/1 (106)*
*telephone-event/8000/1 (126)*
*red/8000/1 (127)*
*WebRtcVideoEngine::WebRtcVideoEngine*
*Warning(webrtcvideoengine.cc <http://webrtcvideoengine.cc>:838):
SetTraceCallback(0x15c51e5c) failed, err=0*
*WebRtcVoiceEngine::Init*
*WebRtc VoiceEngine Version:*
*VoiceEngine 4.1.0*
*Build: svn:Unavailable(issue687) Oct 29 2013 12:59:30 ?*
*Applying audio options: AudioOptions {aec: false, agc: false, ns: true,
hf: true, swap: false, typing: false, conference: false, agc_delta: 0,
experimental_agc: false, experimental_aec: false, aec_dump: false, }*
*Warning(webrtcvoiceengine.cc <http://webrtcvoiceengine.cc>:811):
SetTypingDetectionStatus(0) failed, err=8003*
*Adjusting AGC level from default -3dB to -3dB*
*WebRtc VoiceEngine codecs:*
*opus/48000/2 (111)*
*ISAC/16000/1 (103)*
*G722/16000/1 (9)*
*ILBC/8000/1 (102)*
*PCMU/8000/1 (0)*
*PCMA/8000/1 (8)*
*CN/32000/1 (106)*
*CN/16000/1 (105)*
*CN/8000/1 (13)*
*red/8000/1 (127)*
*telephone-event/8000/1 (126)*
*WebRtcVoiceEngine::Init Done!*
*WebRtcVideoEngine::Init*
*WebRtcVideoEngine::InitVideoEngine*
*WebRtc VideoEngine Version:*
*VideoEngine 3.45.0*
*Build: svn:Unavailable(issue687) Oct 29 2013 13:00:07 ?*
*VideoEngine Init done*
*Applying audio options: AudioOptions {aec: false, agc: false, ns: true,
hf: true, swap: false, typing: false, conference: false, agc_delta: 0,
experimental_agc: false, experimental_aec: false, aec_dump: false, }*
*Warning(webrtcvoiceengine.cc <http://webrtcvoiceengine.cc>:811):
SetTypingDetectionStatus(0) failed, err=8003*
*Adjusting AGC level from default -3dB to -3dB*
*Allowing SCTP data engine.*
*Generating identity.*
*2014-11-06 14:50:01.616 AppRTCDemo[3403:1019125] PC -
setRemoteDescription.*
*Ignored line: a=sendrecv*
*Ignored line: a=rtcp-mux*
*Ignored line: a=rtpmap:111 opus/48000/2*
*Ignored line: a=rtpmap:103 ISAC/16000*
*Ignored line: a=rtpmap:104 ISAC/32000*
*Ignored line: a=rtpmap:0 PCMU/8000*
*Ignored line: a=rtpmap:8 PCMA/8000*
*Ignored line: a=rtpmap:106 CN/32000*
*Ignored line: a=rtpmap:105 CN/16000*
*Ignored line: a=rtpmap:13 CN/8000*
*Ignored line: a=rtpmap:126 telephone-event/8000*
*Ignored line: a=ssrc:1027749449 cname:20vbtzsaI2W/gqQf*
*Ignored line: a=ssrc:1027749449 msid:Ji5HVTyO3vAprIERrnfQh29IcuqBuKvgexr4
7075601e-0ceb-48e3-b13d-ea8a9324f160*
*Ignored line: a=ssrc:1027749449
mslabel:Ji5HVTyO3vAprIERrnfQh29IcuqBuKvgexr4*
*Ignored line: a=ssrc:1027749449
label:7075601e-0ceb-48e3-b13d-ea8a9324f160*
*Ignored line: a=ssrc:3735928559 cname:mixed*
*Ignored line: a=ssrc:3735928559 label:mixedlabelv0*
*Ignored line: a=ssrc:3735928559 msid:mixedmslabel mixedlabelv0*
*Ignored line: a=ssrc:3735928559 mslabel:mixedmslabel*
*Error(webrtcsdp.cc <http://webrtcsdp.cc>:357): Failed to parse: "".
Reason: Failed to parse audio codecs correctly.*
*2014-11-06 14:50:03.852 AppRTCDemo[3403:1019125] SEQ3-Connect button
pressed ...*
*2014-11-06 14:50:03.852 AppRTCDemo[3403:1019125] SEQ4-Setup AppRTC video
view*
*2014-11-06 14:50:03.852 AppRTCDemo[3403:1019276] SDP onFailure.*
*(lldb) *

_______________________________________________
dev mailing list
dev@jitsi.org
Unsubscribe instructions and other list options:
http://lists.jitsi.org/mailman/listinfo/dev


#4

I don't see anything that wrong. You can experiment with an SDP from Jitsi Meet directly -- hard-code it and pass it to setRemoteDescription. If it is accepted, then go and hunt all the little differences between it and the one you generate. If it isn't accepted, there's probably another problem.

To get an SDP offer from Jitsi meet, start a conference and print
"focus.peerconnection.remoteDescription.sdp" from the javascript console.

Hope that helps,
Boris

···

On 06/11/14 21:54, Peter Mycue wrote:

Hello,

I fixed whitespace and changed end of line from \n to \r\n iike the java
code. no change though.

i found that i was not putting these in for video candidates
a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time

i have no lines like this translated from jingle but the sdp from
appspot server sdp does.
c=IN IP4 217.78.97.15
a=rtcp:57026 IN IP4 217.78.97.15

i have ice-frag and ice-pwd before candidates and appspot example has it
after so i switched order.

NOW i get this:
*Error(webrtcsdp.cc <http://webrtcsdp.cc>:357): Failed to parse: "".
Reason: Failed to parse audio codecs correctly.*


#5

That was it! i have been looking at this sdp so long now its hard to see differences anymore.

it now gets past that.

i believe now that i must createAnswer when i get the call back from the sdp creation!

THANKS SO MUCH!

Peter Mycue
pmycue@us.ibm.com
pmycue@gmail.com
704-626-9772

···

On Nov 6, 2014, at 3:09 PM, Emil Ivov <emcho@jitsi.org> wrote:

You are missing tue port number in the audio and video m=lined. I think Chrome sets this to 1 currently but it'supposed to be 9. In other words:

m=audio 1 RTP/SAVPF 111 103 104 0 8 106 105 13 126

--sent from my mobile

On 06 Nov 2014 8:56 PM, "Peter Mycue" <pmycue@gmail.com> wrote:

Hello,

I fixed whitespace and changed end of line from \n to \r\n iike the java code. no change though.

i found that i was not putting these in for video candidates
  a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
  a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time

i have no lines like this translated from jingle but the sdp from appspot server sdp does.
  c=IN IP4 217.78.97.15
  a=rtcp:57026 IN IP4 217.78.97.15

i have ice-frag and ice-pwd before candidates and appspot example has it after so i switched order.

NOW i get this:
Error(webrtcsdp.cc:357): Failed to parse: "". Reason: Failed to parse audio codecs correctly.

IF there is anything else you need to see to help get to the bottom of this, please let me know.

new sdp looks like this
SDP SO FAR IS
v=0

o=- 9135848008193788882 2 IN IP4 127.0.0.1

s=-

t=0 0

a=group:BUNDLE audio video

m=audio RTP/SAVPF 111 103 104 0 8 106 105 13 126

a=candidate:1 1 udp 2130706431 2002:902:37b8:0:0:0:902:37b8 5439 typ host

a=candidate:2 1 udp 2130706431 9.2.55.184 5439 typ host

a=candidate:3 1 udp 2113937151 fe80:0:0:0:e0e1:bee7:27dd:a0a1 5439 typ host

a=candidate:1 2 udp 2130706430 2002:902:37b8:0:0:0:902:37b8 5440 typ host

a=candidate:2 2 udp 2130706430 9.2.55.184 5440 typ host

a=candidate:3 2 udp 2113937150 fe80:0:0:0:e0e1:bee7:27dd:a0a1 5440 typ host

a=ice-ufrag:d2go5

a=ice-pwd:69unsj2efofr85thvjagj85cae

a=mid:audio

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=ssrc:1027749449 cname:20vbtzsaI2W/gqQf

a=ssrc:1027749449 msid:Ji5HVTyO3vAprIERrnfQh29IcuqBuKvgexr4 7075601e-0ceb-48e3-b13d-ea8a9324f160

a=ssrc:1027749449 mslabel:Ji5HVTyO3vAprIERrnfQh29IcuqBuKvgexr4

a=ssrc:1027749449 label:7075601e-0ceb-48e3-b13d-ea8a9324f160

a=ssrc:3735928559 cname:mixed

a=ssrc:3735928559 label:mixedlabelv0

a=ssrc:3735928559 msid:mixedmslabel mixedlabelv0

a=ssrc:3735928559 mslabel:mixedmslabel

a=fingerprint:sha-1 E7:38:F7:2F:12:07:C5:10:CA:3C:21:F9:47:D2:A0:C5:09:51:30:C3

m=video RTP/SAVPF 100 116 117

a=candidate:1 1 udp 2130706431 2002:902:37b8:0:0:0:902:37b8 5443 typ host

a=candidate:2 1 udp 2130706431 9.2.55.184 5443 typ host

a=candidate:3 1 udp 2113937151 fe80:0:0:0:e0e1:bee7:27dd:a0a1 5443 typ host

a=candidate:1 2 udp 2130706430 2002:902:37b8:0:0:0:902:37b8 5444 typ host

a=candidate:2 2 udp 2130706430 9.2.55.184 5444 typ host

a=candidate:3 2 udp 2113937150 fe80:0:0:0:e0e1:bee7:27dd:a0a1 5444 typ host

a=ice-ufrag:3ls50

a=ice-pwd:5sqtfk8bflm9id28qnakcnd0km

a=mid:video

a=extmap:2 urn:ietf:params:rtp-hdrext:toffset

a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time

a=sendrecv

a=rtcp-mux

a=rtpmap:100 VP8/90000

a=rtpmap:116 red/90000

a=rtpmap:117 ulpfec/90000

a=ssrc:3333184907 cname:2t951WLKfY9O8W3h

a=ssrc:3333184907 msid:ABDQ3PByI8HAsSw1db2nRU7rlpzEzCB22Ohe 45ed8da3-32fd-4580-86a5-34233abcb46c

a=ssrc:3333184907 mslabel:ABDQ3PByI8HAsSw1db2nRU7rlpzEzCB22Ohe

a=ssrc:3333184907 label:45ed8da3-32fd-4580-86a5-34233abcb46c

a=ssrc:3735928559 cname:mixed

a=ssrc:3735928559 label:mixedlabelv0

a=ssrc:3735928559 msid:mixedmslabel mixedlabelv0

a=ssrc:3735928559 mslabel:mixedmslabel

a=fingerprint:sha-1 5A:B7:C9:86:40:3A:96:38:D2:41:11:43:76:1C:62:5D:94:F6:20:37

2014-11-06 14:49:59.227 AppRTCDemo[3403:1019125] SEND: <presence type="unavailable"><x xmlns="vcard-temp:x:update"><photo/></x></presence>
2014-11-06 14:49:59.252 AppRTCDemo[3403:1019125] ---------- xmppStreamDidDisconnect: ----------
2014-11-06 14:49:59.258 AppRTCDemo[3403:1019125] UITableView has 2 rows
2014-11-06 14:50:00.139 AppRTCDemo[3403:1019125] SEQ2-Sending CONNECT to room # 98303690
2014-11-06 14:50:00.140 AppRTCDemo[3403:1019125] *** HERE in acceptConnection
Warning(webrtcvoiceengine.cc:360): SetTraceCallback() failed, err=0
WebRtc VoiceEngine codecs:
ISAC/16000/1 (103)
PCMU/8000/1 (0)
PCMA/8000/1 (8)
Warning(webrtcvoiceengine.cc:462): Unexpected codec: PCMU/8000/2 (110)
Warning(webrtcvoiceengine.cc:462): Unexpected codec: PCMA/8000/2 (118)
ILBC/8000/1 (102)
G722/16000/1 (9)
Warning(webrtcvoiceengine.cc:462): Unexpected codec: G722/16000/2 (119)
opus/48000/2 (111)
CN/8000/1 (13)
CN/16000/1 (105)
CN/32000/1 (106)
telephone-event/8000/1 (126)
red/8000/1 (127)
WebRtcVideoEngine::WebRtcVideoEngine
Warning(webrtcvideoengine.cc:838): SetTraceCallback(0x15c51e5c) failed, err=0
WebRtcVoiceEngine::Init
WebRtc VoiceEngine Version:
VoiceEngine 4.1.0
Build: svn:Unavailable(issue687) Oct 29 2013 12:59:30 ?
Applying audio options: AudioOptions {aec: false, agc: false, ns: true, hf: true, swap: false, typing: false, conference: false, agc_delta: 0, experimental_agc: false, experimental_aec: false, aec_dump: false, }
Warning(webrtcvoiceengine.cc:811): SetTypingDetectionStatus(0) failed, err=8003
Adjusting AGC level from default -3dB to -3dB
WebRtc VoiceEngine codecs:
opus/48000/2 (111)
ISAC/16000/1 (103)
G722/16000/1 (9)
ILBC/8000/1 (102)
PCMU/8000/1 (0)
PCMA/8000/1 (8)
CN/32000/1 (106)
CN/16000/1 (105)
CN/8000/1 (13)
red/8000/1 (127)
telephone-event/8000/1 (126)
WebRtcVoiceEngine::Init Done!
WebRtcVideoEngine::Init
WebRtcVideoEngine::InitVideoEngine
WebRtc VideoEngine Version:
VideoEngine 3.45.0
Build: svn:Unavailable(issue687) Oct 29 2013 13:00:07 ?
VideoEngine Init done
Applying audio options: AudioOptions {aec: false, agc: false, ns: true, hf: true, swap: false, typing: false, conference: false, agc_delta: 0, experimental_agc: false, experimental_aec: false, aec_dump: false, }
Warning(webrtcvoiceengine.cc:811): SetTypingDetectionStatus(0) failed, err=8003
Adjusting AGC level from default -3dB to -3dB
Allowing SCTP data engine.
Generating identity.
2014-11-06 14:50:01.616 AppRTCDemo[3403:1019125] PC - setRemoteDescription.
Ignored line: a=sendrecv
Ignored line: a=rtcp-mux
Ignored line: a=rtpmap:111 opus/48000/2
Ignored line: a=rtpmap:103 ISAC/16000
Ignored line: a=rtpmap:104 ISAC/32000
Ignored line: a=rtpmap:0 PCMU/8000
Ignored line: a=rtpmap:8 PCMA/8000
Ignored line: a=rtpmap:106 CN/32000
Ignored line: a=rtpmap:105 CN/16000
Ignored line: a=rtpmap:13 CN/8000
Ignored line: a=rtpmap:126 telephone-event/8000
Ignored line: a=ssrc:1027749449 cname:20vbtzsaI2W/gqQf
Ignored line: a=ssrc:1027749449 msid:Ji5HVTyO3vAprIERrnfQh29IcuqBuKvgexr4 7075601e-0ceb-48e3-b13d-ea8a9324f160
Ignored line: a=ssrc:1027749449 mslabel:Ji5HVTyO3vAprIERrnfQh29IcuqBuKvgexr4
Ignored line: a=ssrc:1027749449 label:7075601e-0ceb-48e3-b13d-ea8a9324f160
Ignored line: a=ssrc:3735928559 cname:mixed
Ignored line: a=ssrc:3735928559 label:mixedlabelv0
Ignored line: a=ssrc:3735928559 msid:mixedmslabel mixedlabelv0
Ignored line: a=ssrc:3735928559 mslabel:mixedmslabel
Error(webrtcsdp.cc:357): Failed to parse: "". Reason: Failed to parse audio codecs correctly.
2014-11-06 14:50:03.852 AppRTCDemo[3403:1019125] SEQ3-Connect button pressed ...
2014-11-06 14:50:03.852 AppRTCDemo[3403:1019125] SEQ4-Setup AppRTC video view
2014-11-06 14:50:03.852 AppRTCDemo[3403:1019276] SDP onFailure.
(lldb)

_______________________________________________
dev mailing list
dev@jitsi.org
Unsubscribe instructions and other list options:
http://lists.jitsi.org/mailman/listinfo/dev
_______________________________________________
dev mailing list
dev@jitsi.org
Unsubscribe instructions and other list options:
http://lists.jitsi.org/mailman/listinfo/dev