Re: Bad RTP pt 104 (expecting 9) + random source warning

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For me it looks like there is only the 1 codec now in the SDP message. But you can look in the logs below and tell me if im right? ;) 

Have you any suggestions for me for my other problem with the sound: 


Sideinfo: Im using the pjlib cross compiled for an raspberry pi 3 and i am using it with Qt. 


WARNING: no real random source present!
Codec: "G722/16000/1" prio: 128
Codec: "PCMA/8000/1" prio: 0
Codec: "PCMU/8000/1" prio: 0
Codec: "GSM/8000/1" prio: 0
Codec: "iLBC/8000/1" prio: 0
Codec: "speex/32000/1" prio: 0
Codec: "speex/8000/1" prio: 0
Codec: "speex/16000/1" prio: 0
Codec: "L16/44100/1" prio: 0
Codec: "L16/44100/2" prio: 0
Codec: "L16/8000/1" prio: 0
Codec: "L16/8000/2" prio: 0
Codec: "L16/16000/1" prio: 0
Codec: "L16/16000/2" prio: 0
Audio Devices available: 8
Device [ 0 ] "default:CARD=ALSA"
Device [ 1 ] "sysdefault:CARD=ALSA"
Device [ 2 ] "dmix:CARD=ALSA,DEV=0"
Device [ 3 ] "dmix:CARD=ALSA,DEV=1"
Device [ 4 ] "hw:CARD=ALSA,DEV=0"
Device [ 5 ] "hw:CARD=ALSA,DEV=1"
Device [ 6 ] "plughw:CARD=ALSA,DEV=0"
Device [ 7 ] "plughw:CARD=ALSA,DEV=1"
*** PJSUA2 STARTED ***
06:59:57.924 sip_endpoint.c .Creating endpoint instance...
06:59:57.924 pjlib .select() I/O Queue created (0x208a138)
06:59:57.924 sip_endpoint.c .Module "mod-msg-print" registered
06:59:57.924 sip_transport. .Transport manager created.
06:59:57.924 pjsua_core.c .PJSUA state changed: NULL --> CREATED
06:59:57.924 sip_endpoint.c .Module "mod-pjsua-log" registered
06:59:57.924 sip_endpoint.c .Module "mod-tsx-layer" registered
06:59:57.924 sip_endpoint.c .Module "mod-stateful-util" registered
06:59:57.924 sip_endpoint.c .Module "mod-ua" registered
06:59:57.924 sip_endpoint.c .Module "mod-100rel" registered
06:59:57.924 sip_endpoint.c .Module "mod-pjsua" registered
06:59:57.924 sip_endpoint.c .Module "mod-invite" registered
06:59:57.990 alsa_dev.c ..ALSA driver found 8 devices
06:59:57.990 alsa_dev.c ..ALSA initialized
06:59:57.990 pjlib ..select() I/O Queue created (0x20b09ac)
06:59:57.997 sip_endpoint.c .Module "mod-evsub" registered
06:59:57.997 sip_endpoint.c .Module "mod-presence" registered
06:59:57.997 sip_endpoint.c .Module "mod-mwi" registered
06:59:57.997 sip_endpoint.c .Module "mod-refer" registered
06:59:57.997 sip_endpoint.c .Module "mod-pjsua-pres" registered
06:59:57.997 sip_endpoint.c .Module "mod-pjsua-im" registered
06:59:57.997 sip_endpoint.c .Module "mod-pjsua-options" registered
06:59:57.997 pjsua_core.c .1 SIP worker threads created
06:59:57.997 pjsua_core.c .pjsua version 2.6 for Linux-4.9.35/armv7l/glibc-2.19 initialized
06:59:57.997 pjsua_core.c .PJSUA state changed: CREATED --> INIT
06:59:57.997 pjsua_aud.c Set sound device: capture=-99, playback=-2
06:59:57.997 pjsua_aud.c Set sound device: capture=-99, playback=-99
06:59:57.997 pjsua_aud.c .Setting null sound device..
06:59:57.997 pjsua_aud.c ..Opening null sound device..
06:59:57.999 pjsua_core.c SIP UDP socket reachable at 192.168.178.42:5060
06:59:57.999 udp0x209a548 SIP UDP transport started, published address is 192.168.178.42:5060
06:59:57.999 pjsua_core.c PJSUA state changed: INIT --> STARTING
06:59:57.999 sip_endpoint.c .Module "mod-unsolicited-mwi" registered
06:59:57.999 pjsua_core.c .PJSUA state changed: STARTING --> RUNNING
06:59:57.999 pjsua_acc.c Adding account: id=sip:doorz-control@xxxxxxxxx
06:59:57.999 pjsua_acc.c .Account sip:doorz-control@xxxxxxxxx added with id 0
06:59:57.999 pjsua_acc.c .Acc 0: setting registration..
06:59:58.002 pjsua_core.c ...TX 504 bytes Request msg REGISTER/cseq=61758 (tdta0x20c43f8) to UDP 192.168.178.1:5060:
REGISTER sip:fritz.box SIP/2.0
Via: SIP/2.0/UDP 192.168.178.42:5060;rport;branch=z9hG4bKPje1dk97R8q3I15dvyocTkdcQqgr4ES.JW
Max-Forwards: 70
From: <sip:doorz-control@xxxxxxxxx>;tag=ENVAypb9UtGAaaD3W8FKNqWiSSKFEYHg
Call-ID: VqR3bDyVbnW.Nu8F6SCeFy18wv5NHHSO
CSeq: 61758 REGISTER
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length: 0


--end msg--
06:59:58.003 pjsua_acc.c ..Acc 0: Registration sent
06:59:58.009 pjsua_core.c .RX 432 bytes Response msg 401/REGISTER/cseq=61758 (rdata0x209bb7c) from UDP 192.168.178.1:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.178.42:5060;rport=5060;branch=z9hG4bKPje1dk97R8q3I15dvyocTkdcQqgr4ES.JW
From: <sip:doorz-control@xxxxxxxxx>;tag=ENVAypb9UtGAaaD3W8FKNqWiSSKFEYHg
To: <sip:doorz-control@xxxxxxxxx>;tag=19989AFA51801DDB
Call-ID: VqR3bDyVbnW.Nu8F6SCeFy18wv5NHHSO
CSeq: 61758 REGISTER
WWW-Authenticate: Digest realm="fritz.box", nonce="03D17C6267BD8920"
User-Agent: FRITZ!OS
Content-Length: 0


--end msg--
06:59:58.009 pjsua_core.c ....TX 663 bytes Request msg REGISTER/cseq=61759 (tdta0x20c43f8) to UDP 192.168.178.1:5060:
REGISTER sip:fritz.box SIP/2.0
Via: SIP/2.0/UDP 192.168.178.42:5060;rport;branch=z9hG4bKPjT5ouchBYqi8vWXHzMqn3Hw9ErM6.SzKx
Max-Forwards: 70
From: <sip:doorz-control@xxxxxxxxx>;tag=ENVAypb9UtGAaa*** Register: code= 200
Start CALL!
MyCall::onCallState
Call-ID: VqR3bDyVbnW.Nu8F6SCeFy18wv5NHHSO
CSeq: 61759 REGISTER
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Authorization: Digest username="doorz-control", realm="fritz.box", nonce="03D17C6267BD8920", uri="sip:fritz.box", response="d6bfb98b8945004cf77268cc99f8584b"
Content-Length: 0


--end msg--
06:59:58.017 pjsua_core.c .RX 698 bytes Response msg 200/REGISTER/cseq=61759 (rdata0x7550169c) from UDP 192.168.178.1:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.178.42:5060;rport=5060;branch=z9hG4bKPjT5ouchBYqi8vWXHzMqn3Hw9ErM6.SzKx
From: <sip:doorz-control@xxxxxxxxx>;tag=ENVAypb9UtGAaaD3W8FKNqWiSSKFEYHg
To: <sip:doorz-control@xxxxxxxxx>;tag=22DE2A38BEF81644
Call-ID: VqR3bDyVbnW.Nu8F6SCeFy18wv5NHHSO
CSeq: 61759 REGISTER
User-Agent: AVM FRITZ!Box Fon WLAN 7390 84.06.83 (Mar 8 2017)
Supported: 100rel,replaces,timer
Allow-Events: telephone-event,refer,reg
Allow: INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE,PUBLISH
Accept: application/sdp, multipart/mixed
Accept-Encoding: identity
Content-Length: 0


--end msg--
06:59:58.017 pjsua_acc.c ....SIP outbound status for acc 0 is not active
06:59:58.018 pjsua_acc.c ....sip:doorz-control@xxxxxxxxx: registration success, status=200 (OK), will re-register in 300 seconds
06:59:58.018 pjsua_acc.c ....Keep-alive timer started for acc 0, destination:192.168.178.1:5060, interval:15s
06:59:58.999 pjsua_aud.c Closing sound device after idle for 1 second(s)
06:59:58.999 pjsua_aud.c .Closing null sound device..
06:59:59.003 pjsua_call.c !Making call with acc #0 to sip:**1@xxxxxxxxx
06:59:59.018 pjsua_aud.c .Set sound device: capture=-99, playback=-99
06:59:59.018 pjsua_aud.c ..Setting null sound device..
06:59:59.018 pjsua_aud.c ...Opening null sound device..
06:59:59.018 pjsua_media.c .Call 0: initializing media..
06:59:59.019 pjsua_media.c ..RTP socket reachable at 192.168.178.42:4000
06:59:59.019 pjsua_media.c ..RTCP socket reachable at 192.168.178.42:4001
06:59:59.019 pjsua_media.c ..Media index 0 selected for audio call 0
06:59:59.022 pjsua_core.c ....TX 870 bytes Request msg INVITE/cseq=10060 (tdta0x20cbc10) to UDP 192.168.178.1:5060:
INVITE sip:**1@xxxxxxxxx SIP/2.0
Via: SIP/2.0/UDP 192.168.178.42:5060;rport;branch=z9hG4bKPjfXj0f0BTz7AIM-YUaweJGxglRroSvjsf
Max-Forwards: 70
Call-ID: jH1tw2KBSeczVda3MoNbKcsViQAFvzVL
CSeq: 10060 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
Content-Type: application/sdp
Content-Length: 277

v=0
o=- 3710811599 3710811599 IN IP4 192.168.178.42
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4000 RTP/AVP 9 96
c=IN IP4 192.168.178.42
b=TIAS:64000
a=rtcp:4001 IN IP4 192.168.178.42
a=sendrecv
a=rtpmap:9 G722/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16

--end msg--
06:59:59.028 pjsua_core.c .RX 420 bytes Response msg 401/INVITE/cseq=10060 (rdata0x7550169c) from UDP 192.168.178.1:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.178.42:5060;rport=5060;branch=z9hG4bKPjfXj0f0BTz7AIM-YUaweJGxglRroSvjsf
From: <sip:doorz-control@xxxxxxxxx>;tag=JtPN-VUUOmhq3.I4c4KkZcjZu-xz.95D
To: <sip:**1@xxxxxxxxx>;tag=AAEAB2A16C04D12D
Call-ID: jH1tw2KBSeczVda3MoNbKcsViQAFvzVL
CSeq: 10060 INVITE
WWW-Authenticate: Digest realm="fritz.box", nonce="1C693C3BF2C2709E"
User-Agent: FRITZ!OS
Content-Length: 0


--end msg--
06:59:59.028 pjsua_core.c ..TX 340 bytes Request msg ACK/cseq=10060 (tdta0x75503bd0) to UDP 192.168.178.1:5060:
ACK sip:**1@xxxxxxxxx SIP/2.0
Via: SIP/2.0/UDP 192.168.178.42:5060;rport;branch=z9hG4bKPjfXj0f0BMyCall::onCallState
MyCall::onCallMediaState
Max-Forwards: 70
Call-ID: jH1tw2KBSeczVda3MoNbKcsViQAFvzVL
CSeq: 10060 ACK
Content-Length: 0


--end msg--
06:59:59.029 pjsua_core.c .......TX 1033 bytes Request msg INVITE/cseq=10061 (tdta0x20cbc10) to UDP 192.168.178.1:5060:
INVITE sip:**1@xxxxxxxxx SIP/2.0
Via: SIP/2.0/UDP 192.168.178.42:5060;rport;branch=z9hG4bKPjzuUpsmNa3R2odjt52f-Zn0ogRbiQ0auT
Max-Forwards: 70
Call-ID: jH1tw2KBSeczVda3MoNbKcsViQAFvzVL
CSeq: 10061 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
Authorization: Digest username="doorz-control", realm="fritz.box", nonce="1C693C3BF2C2709E", uri="sip:**1@xxxxxxxxx", response="9ca5d1aa196c023dbcd967a5d7375bdd"
Content-Type: application/sdp
Content-Length: 277

v=0
o=- 3710811599 3710811599 IN IP4 192.168.178.42
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4000 RTP/AVP 9 96
c=IN IP4 192.168.178.42
b=TIAS:64000
a=rtcp:4001 IN IP4 192.168.178.42
a=sendrecv
a=rtpmap:9 G722/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16

--end msg--
06:59:59.050 pjsua_core.c .RX 365 bytes Response msg 100/INVITE/cseq=10061 (rdata0x7550169c) from UDP 192.168.178.1:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.178.42:5060;rport=5060;branch=z9hG4bKPjzuUpsmNa3R2odjt52f-Zn0ogRbiQ0auT
From: <sip:doorz-control@xxxxxxxxx>;tag=JtPN-VUUOmhq3.I4c4KkZcjZu-xz.95D
Call-ID: jH1tw2KBSeczVda3MoNbKcsViQAFvzVL
CSeq: 10061 INVITE
User-Agent: AVM FRITZ!Box Fon WLAN 7390 84.06.83 (Mar 8 2017)
Content-Length: 0


--end msg--
06:59:59.079 pjsua_core.c .RX 707 bytes Response msg 183/INVITE/cseq=10061 (rdata0x7550169c) from UDP 192.168.178.1:5060:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.178.42:5060;rport=5060;branch=z9hG4bKPjzuUpsmNa3R2odjt52f-Zn0ogRbiQ0auT
From: <sip:doorz-control@xxxxxxxxx>;tag=JtPN-VUUOmhq3.I4c4KkZcjZu-xz.95D
To: <sip:**1@xxxxxxxxx>;tag=CC3E030550BCB9D5
Call-ID: jH1tw2KBSeczVda3MoNbKcsViQAFvzVL
CSeq: 10061 INVITE
User-Agent: AVM FRITZ!Box Fon WLAN 7390 84.06.83 (Mar 8 2017)
Content-Type: application/sdp
Content-Length: 216

v=0
o=user 4919567 4919567 IN IP4 192.168.178.1
s=pjmedia
c=IN IP4 192.168.178.1
t=0 0
m=audio 7078 RTP/AVP 9 96
a=rtpmap:9 G722/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=sendrecv
a=rtcp:7079

--end msg--
06:59:59.079 pjsua_media.c .....Call 0: updating media..
06:59:59.079 pjsua_aud.c ......Audio channel update..
06:59:59.080 strm0x7550836c .......VAD temporarily disabled
06:59:59.080 strm0x7550836c .......Encoder stream started
06:59:59.080 strm0x7550836c .......Decoder stream started
06:59:59.080 pjsua_media.c ......Audio updated, stream #0: G722 (sendrecv)
06:59:59.080 pjsua_aud.c .....Conf connect: 1 --> 0
06:59:59.080 conference.c ......Port 1 (sip:**1@xxxxxxxxx) transmitting to port 0 (Master/sound)
06:59:59.491 stream.c G722 codec used, remote samples per frame detected = 80
06:59:59.718 strm0x7550836c VAD re-enabled
07:00:08.602 pjsua_core.c .RX 803 bytes Response msg 488/INVITE/cseq=10061 (rdata0x7550169c) from UDP 192.168.178.1:5060:
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 192.168.178.42:5060;rport=5060;branch=z9hG4bKPjzuUpsmNa3R2odjt52f-Zn0ogRbiQ0auT
From: <sip:doorz-control@xxxxxxxxx>;tag=JtPN-VUUOmhq3.I4c4KkZcjZu-xz.95D
To: <sip:**1@xxxxxxxxx>;tag=CC3E030550BCB9D5
Call-ID: jH1tw2KBSeczVda3MoNbKcsViQAFvzVL
CSeq: 10061 INVITE
Warning: 399 0.0.0.0 "successful but result empty"
User-Agent: FRITZ!OS
Content-Type: application/sdp
Content-Length: 359

v=0
o=user 4919567 4919568 IN IP4 192.168.178.1
s=call
c=IN IP4 192.168.178.1
t=0 0
m=audioMyCall::onCallState


 
Am 04.08.2017 um 06:55 schrieb Andreas Wehrmann <a.wehrmann@xxxxxxxxxx>:

On 08/04/2017 06:47 AM, Kevin Rombach via pjsip wrote:
I set all other codecs priorities to zero! Like i read in the docs this should be completely disable them? Or is there another way to disable them in the right way?

You're right, I've just checked the docs. Setting the prio to zero _should_ disable them.

http://www.pjsip.org/docs/latest-2/pjmedia/docs/html/group__PJMEDIA__CODEC.htm#gacfc4266d50474b348c8c7a0bf9d54abb

Did you check the message exchange between your app and the Fritzbox again?
If you disabled all but one codec in PJ, you should see a difference in your SDP message from the one you provided previously.



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