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

[Date Prev][Date Next][Thread Prev][Thread Next][Date Index][Thread Index]

 



On 08/04/2017 07:03 AM, Kevin Rombach via pjsip wrote:
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:

 Unable to find default audio device <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/2017-August/020128.html>

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

Sorry no clue, I haven't played around with RPi and PJ yet.

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
Contact: <sip:EEE303552C7E89C15FFEDA99CA2A7@192.168.178.1>
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


Hmm, Fritzbox now rejects the call... this is weird.
I'd say you go ahead and disable all but PCMA/8000 and see if THAT works; just to make sure it works at all...



_______________________________________________
Visit our blog: http://blog.pjsip.org

pjsip mailing list
pjsip@xxxxxxxxxxxxxxx
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org



[Index of Archives]     [Asterisk Users]     [Asterisk App Development]     [Linux ARM Kernel]     [Linux ARM]     [Linux Omap]     [Fedora ARM]     [IETF Annouce]     [Security]     [Bugtraq]     [Linux]     [Linux OMAP]     [Linux MIPS]     [Linux API]
  Powered by Linux