Empathy List Archives

pjsip@lists.pjsip.org

pjsip list

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

Kevin Rombach

Fri, Aug 4, 2017 5:48 AM

Yeah im setting the nullDevice for call tests because i can not make a call when the default audio device is not found :(

ep.audDevManager().setCaptureDev(-99);
ep.audDevManager().setPlaybackDev(-99);

when i set the audio device like this:

ep.audDevManager().setCaptureDev(0);
ep.audDevManager().setPlaybackDev(-99);

im getting the error:

"pjsua_call_make_call(acc.getId(), &pj_dst_uri, param.p_opt, this, param.p_msg_data, &id) error: Unable to find default audio device (PJMEDIA_EAUD_NODEFDEV) (status=420006) [../src/pjsua2/call.cpp:490]”

the thing is, i have currently no capture device on my RPi. I first have to order a USB soundcard. Thats why i try to just enable the default playback device and set the capture device to null.

Am 04.08.2017 um 07:37 schrieb Andreas Wehrmann a.wehrmann@yandex.com:

On 08/04/2017 07:24 AM, Kevin Rombach via pjsip wrote:

Okay. But lets say i would not be on the RPi but just on linux and i would have the same problem with the “No default audio device” any idea where this could come from?

Have you checked my last logs? Is the codec thing correct now?

Yes I did, see below; PJ correctly offers G.722 only.
I was wondering why the Fritzbox is now rejecting your call and suggested you enable only PCMA/8000 to see if it works at all.

I checked your logs again and found that only the "NULL sound device" is connected to your confbridge.
So it looks like there is no "real" sound port connected to any confbridge port,
which would explain the lack of audio:

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..

Are you telling PJSUA to use the NULL device when setting it up?
See:
http://www.pjsip.org/docs/latest-2/pjsip/docs/html/group__PJSUA__LIB__MEDIA.htm#ga2e6cb631c6ca40d30973cc5ebeaba255

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@fritz.box;tag=JtPN-VUUOmhq3.I4c4KkZcjZu-xz.95D
To: sip:**1@fritz.box;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@fritz.box) 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@fritz.box;tag=JtPN-VUUOmhq3.I4c4KkZcjZu-xz.95D
To: sip:**1@fritz.box;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@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org

Yeah im setting the nullDevice for call tests because i can not make a call when the default audio device is not found :( ep.audDevManager().setCaptureDev(-99); ep.audDevManager().setPlaybackDev(-99); when i set the audio device like this: ep.audDevManager().setCaptureDev(0); ep.audDevManager().setPlaybackDev(-99); im getting the error: "pjsua_call_make_call(acc.getId(), &pj_dst_uri, param.p_opt, this, param.p_msg_data, &id) error: Unable to find default audio device (PJMEDIA_EAUD_NODEFDEV) (status=420006) [../src/pjsua2/call.cpp:490]” the thing is, i have currently no capture device on my RPi. I first have to order a USB soundcard. Thats why i try to just enable the default playback device and set the capture device to null. > Am 04.08.2017 um 07:37 schrieb Andreas Wehrmann <a.wehrmann@yandex.com>: > > > On 08/04/2017 07:24 AM, Kevin Rombach via pjsip wrote: >> Okay. But lets say i would not be on the RPi but just on linux and i would have the same problem with the “No default audio device” any idea where this could come from? >> >> >> Have you checked my last logs? Is the codec thing correct now? >> >> >>> > > Yes I did, see below; PJ correctly offers G.722 only. > I was wondering why the Fritzbox is now rejecting your call and suggested you enable only PCMA/8000 to see if it works at all. > > I checked your logs again and found that only the "NULL sound device" is connected to your confbridge. > So it looks like there is no "real" sound port connected to any confbridge port, > which would explain the lack of audio: > > 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.. > > Are you telling PJSUA to use the NULL device when setting it up? > See: > http://www.pjsip.org/docs/latest-2/pjsip/docs/html/group__PJSUA__LIB__MEDIA.htm#ga2e6cb631c6ca40d30973cc5ebeaba255 > > >>>> 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@fritz.box>;tag=JtPN-VUUOmhq3.I4c4KkZcjZu-xz.95D >>>> To: <sip:**1@fritz.box>;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@fritz.box) 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@fritz.box>;tag=JtPN-VUUOmhq3.I4c4KkZcjZu-xz.95D >>>> To: <sip:**1@fritz.box>;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@lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org

Andreas Wehrmann

Fri, Aug 4, 2017 7:34 AM

On 08/04/2017 07:48 AM, Kevin Rombach via pjsip wrote:

Yeah im setting the nullDevice for call tests because i can not make a call when the default audio device is not found :(

  ep.audDevManager().setCaptureDev(-99);
  ep.audDevManager().setPlaybackDev(-99);

when i set the audio device like this:

  ep.audDevManager().setCaptureDev(0);
  ep.audDevManager().setPlaybackDev(-99);

im getting the error:

the thing is, i have currently no capture device on my RPi. I first have to order a USB soundcard. Thats why i try to just enable the default playback device and set the capture device to null.

Hmm, I'm not familiar with the C++ API, but what you might try is
creating a "sound player" port explicitly and then connect your call to it:

You need to create a player port, using
pjmedia_snd_port_create_player(); this gives you a pjmedia_snd_port.
http://www.pjsip.org/docs/latest-2/pjmedia/docs/html/group__PJMED__SND__PORT.htm#gaa67792594c2b15e22f957f12e90cd206

Create a splitter/combiner port, using pjmedia_splitcomb_create();
register it to the conference bridge; this gives you a pjmedia_port:
http://www.pjsip.org/docs/latest-2/pjmedia/docs/html/group__PJMEDIA__SPLITCOMB.htm#ga4cc0abefb5b62ee8d88f875994e548dd

Now connect the sound player port to the splitter/combiner port using
pjmedia_snd_port_connect():
http://www.pjsip.org/docs/latest-2/pjmedia/docs/html/group__PJMED__SND__PORT.htm#ga046156b765a34e6c640b0534e6b21f9c

In your call confirmation callback, you should now be able to connect
the confport of the call to the splitter/combiner port which is
connected to your audio output.

On 08/04/2017 07:48 AM, Kevin Rombach via pjsip wrote: > Yeah im setting the nullDevice for call tests because i can not make a call when the default audio device is not found :( > > ep.audDevManager().setCaptureDev(-99); > ep.audDevManager().setPlaybackDev(-99); > > when i set the audio device like this: > > ep.audDevManager().setCaptureDev(0); > ep.audDevManager().setPlaybackDev(-99); > > im getting the error: > > "pjsua_call_make_call(acc.getId(), &pj_dst_uri, param.p_opt, this, param.p_msg_data, &id) error: Unable to find default audio device (PJMEDIA_EAUD_NODEFDEV) (status=420006) [../src/pjsua2/call.cpp:490]” > > > the thing is, i have currently no capture device on my RPi. I first have to order a USB soundcard. Thats why i try to just enable the default playback device and set the capture device to null. > Hmm, I'm not familiar with the C++ API, but what you might try is creating a "sound player" port explicitly and then connect your call to it: You need to create a player port, using pjmedia_snd_port_create_player(); this gives you a pjmedia_snd_port. http://www.pjsip.org/docs/latest-2/pjmedia/docs/html/group__PJMED__SND__PORT.htm#gaa67792594c2b15e22f957f12e90cd206 Create a splitter/combiner port, using pjmedia_splitcomb_create(); register it to the conference bridge; this gives you a pjmedia_port: http://www.pjsip.org/docs/latest-2/pjmedia/docs/html/group__PJMEDIA__SPLITCOMB.htm#ga4cc0abefb5b62ee8d88f875994e548dd Now connect the sound player port to the splitter/combiner port using pjmedia_snd_port_connect(): http://www.pjsip.org/docs/latest-2/pjmedia/docs/html/group__PJMED__SND__PORT.htm#ga046156b765a34e6c640b0534e6b21f9c In your call confirmation callback, you should now be able to connect the confport of the call to the splitter/combiner port which is connected to your audio output.

s.marek@avm.de

Wed, Sep 13, 2017 12:38 PM

Hi Kevin,

at first: **1 calls the a/b interface for analog phones - G.722 is not
support on this port. That's why the UPDATE is rejected.

You have 2 options:

If you don't need G.722, simply disable it and everything will work
smoothly.
If you want your installation to support G.722, you need to patch PJSIP
to get it up and running with FRITZ!OS.

A long time ago, the PJSIP implementation was questioned on this list:
http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/2013-August/016638.html
Unfortunately no one responded. The content of the provided patch is still
valid, although it cannot be automatically applied using the patch tool
anymore. Too much changes since 2013.

To show you the technical background, I'll cite parts of the two important
requests:

SIP/2.0 183 Session Progress

...

From: sip:control@fritz.box;tag=ZOKYABGChnZHwr7GpEuIJbxhc7GeLr6-

...

o=user 15920484 15920484 IN IP4 192.168.178.1

...

m=audio 7082 RTP/AVP 9 104 0 8 96

FRITZ!OS always "rings" with all supported codecs.

SIP/2.0 200 OK

...

From: sip:control@fritz.box;tag=ZOKYABGChnZHwr7GpEuIJbxhc7GeLr6-

...

o=user 15920484 15920485 IN IP4 192.168.178.1

...

m=audio 7082 RTP/AVP 104 0 8 96

This is the way that FRITZ!OS signals a codec-list change. The tag remains
the same (it's the same context), but the SDP session version is increased
(second number in the "o"-line).

For some reason PJSIP requires a change of the tag parameter, although
nothing is forked here. See the link above.

You can still apply the patch from 2013 by hand, it's pretty easy.

Good luck,
Sebastian.

"pjsip" pjsip-bounces@lists.pjsip.org schrieb am 02.08.2017 17:34:04:

Von: Kevin Rombach via pjsip pjsip@lists.pjsip.org
An: pjsip@lists.pjsip.org
Kopie: Kevin Rombach kevinrombach@googlemail.com
Datum: 02.08.2017 17:35
Betreff: [pjsip] Bad RTP pt 104 (expecting 9) + random source warning
Gesendet von: "pjsip" pjsip-bounces@lists.pjsip.org

Hey there,

i have the “Bad RTP Problem”. Like i researched for now my problem
seems to be that my FritzBox is trying to use the iLBC coded but my
Raspberry Pi3 with PJSUA2 V2.6 is expecting G722 coded. Why is my
PJSUA not adapting to the coded which is transmitted from the
FritzBox? Is there a way to enable codec changing depending on the
received codec somewhere?

And another thing: Im getting the WARNING related to the random
source below when i start my programm. How can i fix the random source?

Greetz and thanks!

WARNING: no real random source present!

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 ***

Codec: "speex/16000/1" prio: 130
Codec: "speex/8000/1" prio: 129
Codec: "speex/32000/1" prio: 128
Codec: "iLBC/8000/1" prio: 128
Codec: "GSM/8000/1" prio: 128
Codec: "PCMU/8000/1" prio: 128
Codec: "PCMA/8000/1" prio: 128
Codec: "G722/16000/1" prio: 128
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

08:03:48.315 os_core_unix.c !pjlib 2.6 for POSIX initialized
08:03:48.317 sip_endpoint.c .Creating endpoint instance...
08:03:48.317 pjlib .select() I/O Queue created (0x1b7a138)
08:03:48.317 sip_endpoint.c .Module "mod-msg-print" registered
08:03:48.317 sip_transport. .Transport manager created.
08:03:48.317 pjsua_core.c .PJSUA state changed: NULL --> CREATED
08:03:48.317 sip_endpoint.c .Module "mod-pjsua-log" registered
08:03:48.317 sip_endpoint.c .Module "mod-tsx-layer" registered
08:03:48.317 sip_endpoint.c .Module "mod-stateful-util" registered
08:03:48.317 sip_endpoint.c .Module "mod-ua" registered
08:03:48.317 sip_endpoint.c .Module "mod-100rel" registered
08:03:48.317 sip_endpoint.c .Module "mod-pjsua" registered
08:03:48.317 sip_endpoint.c .Module "mod-invite" registered
08:03:48.383 alsa_dev.c ..ALSA driver found 8 devices
08:03:48.383 alsa_dev.c ..ALSA initialized
08:03:48.383 pjlib ..select() I/O Queue created (0x1ba09ac)
08:03:48.390 sip_endpoint.c .Module "mod-evsub" registered
08:03:48.390 sip_endpoint.c .Module "mod-presence" registered
08:03:48.390 sip_endpoint.c .Module "mod-mwi" registered
08:03:48.390 sip_endpoint.c .Module "mod-refer" registered
08:03:48.390 sip_endpoint.c .Module "mod-pjsua-pres" registered
08:03:48.390 sip_endpoint.c .Module "mod-pjsua-im" registered
08:03:48.390 sip_endpoint.c .Module "mod-pjsua-options" registered
08:03:48.391 pjsua_core.c .1 SIP worker threads created
08:03:48.391 pjsua_core.c .pjsua version 2.6 for Linux-4.9.35/
armv7l/glibc-2.19 initialized
08:03:48.391 pjsua_core.c .PJSUA state changed: CREATED --> INIT
08:03:48.391 pjsua_aud.c Setting null sound device..
08:03:48.391 pjsua_aud.c .Opening null sound device..
08:03:48.392 pjsua_core.c SIP UDP socket reachable at

192.168.178.42:5060

08:03:48.392 udp0x1b8a548 SIP UDP transport started, published
address is 192.168.178.42:5060
08:03:48.392 pjsua_core.c PJSUA state changed: INIT --> STARTING
08:03:48.392 sip_endpoint.c .Module "mod-unsolicited-mwi" registered
08:03:48.392 pjsua_core.c .PJSUA state changed: STARTING --> RUNNING
08:03:48.392 pjsua_acc.c Adding account: id=sip:control@fritz.box
08:03:48.392 pjsua_acc.c .Account sip:control@fritz.box added with id 0
08:03:48.392 pjsua_acc.c .Acc 0: setting registration..
08:03:48.394 pjsua_core.c ...TX 504 bytes Request msg REGISTER/
cseq=12711 (tdta0x1bb43f8) 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=z9hG4bKPjgDk237BB0Z1As4Djx4OjJ6Ib5OY-hidy
Max-Forwards: 70
From: sip:control@fritz.box;tag=e4CKomYR-BUoZgnDJnWct4GrGq6noRaw
To: sip:control@fritz.box
Call-ID: JFKgSk6EBSp.rT5Ngn2kpmBkuYfUBhMW
CSeq: 12711 REGISTER
Contact: sip:control@192.168.178.42:5060;ob
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE,
NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length: 0

--end msg--
08:03:48.394 pjsua_acc.c ..Acc 0: Registration sent
08:03:48.402 pjsua_core.c .RX 432 bytes Response msg 401/REGISTER/
cseq=12711 (rdata0x1b8bb7c) 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=z9hG4bKPjgDk237BB0Z1As4Djx4OjJ6Ib5OY-hidy
From: sip:control@fritz.box;tag=e4CKomYR-BUoZgnDJnWct4GrGq6noRaw
To: sip:control@fritz.box;tag=80CA5F3576C71F79
Call-ID: JFKgSk6EBSp.rT5Ngn2kpmBkuYfUBhMW
CSeq: 12711 REGISTER
WWW-Authenticate: Digest realm="fritz.box", nonce="75A1D3FC1DE38C16"
User-Agent: FRITZ!OS
Content-Length: 0

--end msg--
08:03:48.403 pjsua_core.c ....TX 663 bytes Request msg REGISTER/
cseq=12712 (tdta0x1bb43f8) 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=z9hG4bKPjXHEmq8A.BiZahhKpc.QOS6NYVR-.ZkTY
Max-Forwards: 70
From: sip:control@fritz.box;tag=e4CKomYR-BUoZgnDJnWct4GrGq6noRaw
To: sip:control@fritz.box
Call-ID: JFKgSk6EBSp.rT5Ngn2kpmBkuYfUBhMW
CSeq: 12712 REGISTER
Contact: <sip:control*** Register: code= 200
Start CALL!
MyCall::onCallState
MyCall::onCallState
MyCall::onCallMediaState
@192.168.178.42:5060;ob>
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE,
NOTIFY, REFER, MESSAGE, OPTIONS
Authorization: Digest username="control", realm="fritz.box",
nonce="75A1D3FC1DE38C16", uri="sip:fritz.box",
response="d65dee7dec9d8b1a160f352bd234f602"
Content-Length: 0

--end msg--
08:03:48.410 pjsua_core.c .RX 698 bytes Response msg 200/REGISTER/
cseq=12712 (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=z9hG4bKPjXHEmq8A.BiZahhKpc.QOS6NYVR-.ZkTY
From: sip:control@fritz.box;tag=e4CKomYR-BUoZgnDJnWct4GrGq6noRaw
To: sip:control@fritz.box;tag=7CC5E4EDC680E987
Call-ID: JFKgSk6EBSp.rT5Ngn2kpmBkuYfUBhMW
CSeq: 12712 REGISTER
Contact: sip:control@192.168.178.42:5060;ob;expires=300
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--
08:03:48.410 pjsua_acc.c ....SIP outbound status for acc 0 is not active
08:03:48.410 pjsua_acc.c ....sip:control@fritz.box: registration
success, status=200 (OK), will re-register in 300 seconds
08:03:48.410 pjsua_acc.c ....Keep-alive timer started for acc 0,
destination:192.168.178.1:5060, interval:15s
08:03:49.392 pjsua_aud.c Closing sound device after idle for 1 second(s)
08:03:49.392 pjsua_aud.c .Closing null sound device..
08:03:58.395 pjsua_call.c !Making call with acc #0 to sip:**1@fritz.box
08:03:58.395 pjsua_aud.c .Set sound device: capture=-99, playback=-99
08:03:58.395 pjsua_aud.c ..Setting null sound device..
08:03:58.395 pjsua_aud.c ...Opening null sound device..
08:03:58.395 pjsua_media.c .Call 0: initializing media..
08:03:58.396 pjsua_media.c ..RTP socket reachable at 192.168.178.42:4000
08:03:58.396 pjsua_media.c ..RTCP socket reachable at

192.168.178.42:4001

08:03:58.396 pjsua_media.c ..Media index 0 selected for audio call 0
08:03:58.399 pjsua_core.c ....TX 1071 bytes Request msg INVITE/
cseq=5320 (tdta0x1bb8918) to UDP 192.168.178.1:5060:
INVITE sip:**1@fritz.box SIP/2.0
Via: SIP/2.0/UDP 192.168.178.42:
5060;rport;branch=z9hG4bKPjoVffQxnRns2M6dRNdVofacQIBB1MU0jm
Max-Forwards: 70
From: sip:control@fritz.box;tag=ZOKYABGChnZHwr7GpEuIJbxhc7GeLr6-
To: sip:**1@fritz.box
Contact: sip:control@192.168.178.42:5060;ob
Call-ID: ImFJX4NogfS.aPyhSunNEulY6K8fdePa
CSeq: 5320 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: 479

v=0
o=- 3710469838 3710469838 IN IP4 192.168.178.42
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4000 RTP/AVP 104 98 97 99 3 0 8 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:104 iLBC/8000
a=fmtp:104 mode=30
a=rtpmap:98 speex/16000
a=rtpmap:97 speex/8000
a=rtpmap:99 speex/32000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16

--end msg--
08:03:58.405 pjsua_core.c .RX 419 bytes Response msg 401/INVITE/
cseq=5320 (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=z9hG4bKPjoVffQxnRns2M6dRNdVofacQIBB1MU0jm
From: sip:control@fritz.box;tag=ZOKYABGChnZHwr7GpEuIJbxhc7GeLr6-
To: sip:**1@fritz.box;tag=A7313C6763DD7B57
Call-ID: ImFJX4NogfS.aPyhSunNEulY6K8fdePa
CSeq: 5320 INVITE
WWW-Authenticate: Digest realm="fritz.box", nonce="A7103BB430D7AE63"
User-Agent: FRITZ!OS
Content-Length: 0

--end msg--
08:03:58.405 pjsua_core.c ..TX 339 bytes Request msg ACK/cseq=5320
(tdta0x75503bd0) to UDP 192.168.178.1:5060:
ACK sip:**1@fritz.box SIP/2.0
Via: SIP/2.0/UD
P

192.168.178.42:5060;rport;branch=z9hG4bKPjoVffQxnRns2M6dRNdVofacQIBB1MU0jm

Max-Forwards: 70
From: sip:control@fritz.box;tag=ZOKYABGChnZHwr7GpEuIJbxhc7GeLr6-
To: sip:**1@fritz.box;tag=A7313C6763DD7B57
Call-ID: ImFJX4NogfS.aPyhSunNEulY6K8fdePa
CSeq: 5320 ACK
Content-Length: 0

--end msg--
08:03:58.405 pjsua_core.c .......TX 1234 bytes Request msg INVITE/
cseq=5321 (tdta0x1bb8918) to UDP 192.168.178.1:5060:
INVITE sip:**1@fritz.box SIP/2.0
Via: SIP/2.0/UDP 192.168.178.42:
5060;rport;branch=z9hG4bKPjs4xNuSbT6cqjpNaEz-LvtSgHoBqC7Ax3
Max-Forwards: 70
From: sip:control@fritz.box;tag=ZOKYABGChnZHwr7GpEuIJbxhc7GeLr6-
To: sip:**1@fritz.box
Contact: sip:control@192.168.178.42:5060;ob
Call-ID: ImFJX4NogfS.aPyhSunNEulY6K8fdePa
CSeq: 5321 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="control", realm="fritz.box",
nonce="A7103BB430D7AE63", uri="sip:**1@fritz.box",
response="1b7bf1bda6e96088bdf72820185ee781"
Content-Type: application/sdp
Content-Length: 479

--end msg--
08:03:58.425 pjsua_core.c .RX 364 bytes Response msg 100/INVITE/
cseq=5321 (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=z9hG4bKPjs4xNuSbT6cqjpNaEz-LvtSgHoBqC7Ax3
From: sip:control@fritz.box;tag=ZOKYABGChnZHwr7GpEuIJbxhc7GeLr6-
To: sip:**1@fritz.box
Call-ID: ImFJX4NogfS.aPyhSunNEulY6K8fdePa
CSeq: 5321 INVITE
User-Agent: AVM FRITZ!Box Fon WLAN 7390 84.06.83 (Mar 8 2017)
Content-Length: 0

--end msg--
08:03:58.458 pjsua_core.c .RX 804 bytes Response msg 183/INVITE/
cseq=5321 (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=z9hG4bKPjs4xNuSbT6cqjpNaEz-LvtSgHoBqC7Ax3
From: sip:control@fritz.box;tag=ZOKYABGChnZHwr7GpEuIJbxhc7GeLr6-
To: sip:**1@fritz.box;tag=F723EFB025BCF533
Call-ID: ImFJX4NogfS.aPyhSunNEulY6K8fdePa
CSeq: 5321 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: 314

v=0
o=user 15920484 15920484 IN IP4 192.168.178.1
s=pjmedia
c=IN IP4 192.168.178.1
t=0 0
m=audio 7082 RTP/AVP 9 104 0 8 96
a=rtpmap:104 iLBC/8000
a=fmtp:104 mode=30
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=sendrecv
a=rtcp:7083

--end msg--
08:03:58.458 pjsua_media.c .....Call 0: updating media..
08:03:58.458 pjsua_aud.c ......Audio channel update..
08:03:58.459 strm0x75507a64 .......VAD temporarily disabled
08:03:58.459 strm0x75507a64 .......Encoder stream started
08:03:58.459 strm0x75507a64 .......Decoder stream started
08:03:58.459 pjsua_media.c ......Audio updated, stream #0: G722

(sendrecv)

08:03:58.459 pjsua_aud.c .....Conf connect: 1 --> 0
08:03:58.459 conference.c ......Port 1 (sip:**1@fritz.box)
transmitting to port 0 (Master/sound)
08:03:58.459 pjsua_aud.c .....Conf connect: 0 --> 1
08:03:58.459 conference.c ......Port 0 (Master/sound) transmitting to

port 1 (

sip:**1@fritz.box)
08:03:58.885 stream.c G722 codec used, remote samples per frame detected

= 80

08:03:59.095 strm0x75507a64 VAD re-enabled
08:04:00.870 pjsua_core.c .RX 1050 bytes Response msg 200/INVITE/
cseq=5321 (rdata0x7550169c) from UDP 192.168.178.1:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168MyCall::onCallState
MyCall::onCallState
.178.42:5060;rport=5060;branch=z9hG4bKPjs4xNuSbT6cqjpNaEz-LvtSgHoBqC7Ax3
From: sip:control@fritz.box;tag=ZOKYABGChnZHwr7GpEuIJbxhc7GeLr6-
To: sip:**1@fritz.box;tag=F723EFB025BCF533
Call-ID: ImFJX4NogfS.aPyhSunNEulY6K8fdePa
CSeq: 5321 INVITE
Contact: sip:EEE303552C7E89C15FFEDA99CA2A7@192.168.178.1
Session-Expires: 1800;refresher=uac
Min-SE: 90
User-Agent: AVM FRITZ!Box Fon WLAN 7390 84.06.83 (Mar 8 2017)
Supported: 100rel,replaces,timer
Allow-Events: telephone-event,refer
Allow:

INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE,PUBLISH

Content-Type: application/sdp
Accept: application/sdp, multipart/mixed
Accept-Encoding: identity
Content-Length: 290

v=0
o=user 15920484 15920485 IN IP4 192.168.178.1
s=pjmedia
c=IN IP4 192.168.178.1
t=0 0
m=audio 7082 RTP/AVP 104 0 8 96
a=rtpmap:104 iLBC/8000
a=fmtp:104 mode=30
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=sendrecv
a=rtcp:7083

--end msg--
08:04:00.871 inv0x1bb445c ....SDP negotiation done, message body is

ignored

08:04:00.871 pjsua_core.c .....TX 369 bytes Request msg ACK/
cseq=5321 (tdta0x7550cac8) to UDP 192.168.178.1:5060:
ACK sip:EEE303552C7E89C15FFEDA99CA2A7@192.168.178.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.178.42:
5060;rport;branch=z9hG4bKPjZTvu2N7Ts4OG31LGeIs5XeeUDCmY5Iap
Max-Forwards: 70
From: sip:control@fritz.box;tag=ZOKYABGChnZHwr7GpEuIJbxhc7GeLr6-
To: sip:**1@fritz.box;tag=F723EFB025BCF533
Call-ID: ImFJX4NogfS.aPyhSunNEulY6K8fdePa
CSeq: 5321 ACK
Content-Length: 0

--end msg--
08:04:00.871 pjsua_call.c .Call 0 sending UPDATE for updating media
session to use only one codec
08:04:00.872 pjsua_core.c ....TX 836 bytes Request msg UPDATE/
cseq=5322 (tdta0x75510b58) to UDP 192.168.178.1:5060:
UPDATE sip:EEE303552C7E89C15FFEDA99CA2A7@192.168.178.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.178.42:5060;rport;branch=z9hG4bKPjOGaH3-
srAqv5CzFr.zdLzHaKV90z789N
Max-Forwards: 70
From: sip:control@fritz.box;tag=ZOKYABGChnZHwr7GpEuIJbxhc7GeLr6-
To: sip:**1@fritz.box;tag=F723EFB025BCF533
Contact: sip:control@192.168.178.42:5060;ob
Call-ID: ImFJX4NogfS.aPyhSunNEulY6K8fdePa
CSeq: 5322 UPDATE
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800;refresher=uac
Min-SE: 90
Content-Type: application/sdp
Content-Length: 277

v=0
o=- 3710469838 3710469839 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=rtpmap:9 G722/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=sendrecv

--end msg--
08:04:00.890 pjsua_core.c .RX 356 bytes Response msg 488/UPDATE/
cseq=5322 (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=z9hG4bKPjOGaH3-srAqv5CzFr.zdLzHaKV90z789N
From: sip:control@fritz.box;tag=ZOKYABGChnZHwr7GpEuIJbxhc7GeLr6-
To: sip:**1@fritz.box;tag=F723EFB025BCF533
Call-ID: ImFJX4NogfS.aPyhSunNEulY6K8fdePa
CSeq: 5322 UPDATE
User-Agent: FRITZ!OS
Content-Length: 0

--end msg—

08:04:00.951 strm0x75507a64 Bad RTP pt 104 (expecting 9)
08:04:00.983 strm0x75507a64 Bad RTP pt 104 (expecting 9)
08:04:01.010 strm0x75507a64 Bad RTP pt 104 (expecting 9)
08:04:01.039 strm0x75507a64 Bad RTP pt 104 (expecting 9)
08:04:01.072 strm0x75507a64 Bad RTP pt 104 (expecting 9)
08:04:01.103 strm0x75507a64 Bad RTP pt 104 (expecting 9)
08:04:01.127 strm0x75507a64 Bad RTP pt 104 (expecting 9)
08:04:01.159 strm0x75507a64 Bad RTP pt 104 (expecting 9)
08:04:01.192 strm0x75507a64 Bad RTP pt 104 (expecting 9)
08:04:01.223 strm0x75507a64 Bad RTP pt 104 (expecting 9)
08:04:01.247 strm0x75507a64 Bad RTP pt 104 (expecting 9)
08:04:01.280 strm0x75507a64 Bad RTP pt 104 (expecting 9)
08:04:01.311 strm0x75507a64 Bad RTP pt 104 (expecting 9)
08:04:01.343 strm0x75507a64 Bad RTP pt 104 (expecting 9)
08:04:01.367 strm0x75507a64 Bad RTP pt 104 (expecting 9)
08:04:01.399 strm0x75507a64 Bad RTP pt 104 (expecting 9)
08:04:01.431 strm0x75507a64 Bad RTP pt 104 (expecting 9)
08:04:01.464 strm0x75507a64 Bad RTP pt 104 (expecting 9)
08:04:01.487 strm0x75507a64 Bad RTP pt 104 (expecting 9)
08:04:01.519 strm0x75507a64 Bad RTP pt 104 (expecting 9)
08:04:01.551 strm0x75507a64 Bad RTP pt 104 (expecting 9)

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

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

Hi Kevin, at first: **1 calls the a/b interface for analog phones - G.722 is not support on this port. That's why the UPDATE is rejected. You have 2 options: 1) If you don't need G.722, simply disable it and everything will work smoothly. 2) If you want your installation to support G.722, you need to patch PJSIP to get it up and running with FRITZ!OS. A long time ago, the PJSIP implementation was questioned on this list: http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/2013-August/016638.html Unfortunately no one responded. The content of the provided patch is still valid, although it cannot be automatically applied using the patch tool anymore. Too much changes since 2013. To show you the technical background, I'll cite parts of the two important requests: > SIP/2.0 183 Session Progress ... > From: <sip:control@fritz.box>;tag=ZOKYABGChnZHwr7GpEuIJbxhc7GeLr6- ... > > o=user 15920484 15920484 IN IP4 192.168.178.1 ... > m=audio 7082 RTP/AVP 9 104 0 8 96 FRITZ!OS always "rings" with all supported codecs. > SIP/2.0 200 OK ... > From: <sip:control@fritz.box>;tag=ZOKYABGChnZHwr7GpEuIJbxhc7GeLr6- ... > > o=user 15920484 15920485 IN IP4 192.168.178.1 ... > m=audio 7082 RTP/AVP 104 0 8 96 This is the way that FRITZ!OS signals a codec-list change. The tag remains the same (it's the same context), but the SDP session version is increased (second number in the "o"-line). For some reason PJSIP requires a change of the tag parameter, although nothing is forked here. See the link above. You can still apply the patch from 2013 by hand, it's pretty easy. Good luck, Sebastian. "pjsip" <pjsip-bounces@lists.pjsip.org> schrieb am 02.08.2017 17:34:04: > Von: Kevin Rombach via pjsip <pjsip@lists.pjsip.org> > An: pjsip@lists.pjsip.org > Kopie: Kevin Rombach <kevinrombach@googlemail.com> > Datum: 02.08.2017 17:35 > Betreff: [pjsip] Bad RTP pt 104 (expecting 9) + random source warning > Gesendet von: "pjsip" <pjsip-bounces@lists.pjsip.org> > > Hey there, > > i have the “Bad RTP Problem”. Like i researched for now my problem > seems to be that my FritzBox is trying to use the iLBC coded but my > Raspberry Pi3 with PJSUA2 V2.6 is expecting G722 coded. Why is my > PJSUA not adapting to the coded which is transmitted from the > FritzBox? Is there a way to enable codec changing depending on the > received codec somewhere? > > And another thing: Im getting the WARNING related to the random > source below when i start my programm. How can i fix the random source? > > Greetz and thanks! > > WARNING: no real random source present! > > 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 *** > > Codec: "speex/16000/1" prio: 130 > Codec: "speex/8000/1" prio: 129 > Codec: "speex/32000/1" prio: 128 > Codec: "iLBC/8000/1" prio: 128 > Codec: "GSM/8000/1" prio: 128 > Codec: "PCMU/8000/1" prio: 128 > Codec: "PCMA/8000/1" prio: 128 > Codec: "G722/16000/1" prio: 128 > 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 > > 08:03:48.315 os_core_unix.c !pjlib 2.6 for POSIX initialized > 08:03:48.317 sip_endpoint.c .Creating endpoint instance... > 08:03:48.317 pjlib .select() I/O Queue created (0x1b7a138) > 08:03:48.317 sip_endpoint.c .Module "mod-msg-print" registered > 08:03:48.317 sip_transport. .Transport manager created. > 08:03:48.317 pjsua_core.c .PJSUA state changed: NULL --> CREATED > 08:03:48.317 sip_endpoint.c .Module "mod-pjsua-log" registered > 08:03:48.317 sip_endpoint.c .Module "mod-tsx-layer" registered > 08:03:48.317 sip_endpoint.c .Module "mod-stateful-util" registered > 08:03:48.317 sip_endpoint.c .Module "mod-ua" registered > 08:03:48.317 sip_endpoint.c .Module "mod-100rel" registered > 08:03:48.317 sip_endpoint.c .Module "mod-pjsua" registered > 08:03:48.317 sip_endpoint.c .Module "mod-invite" registered > 08:03:48.383 alsa_dev.c ..ALSA driver found 8 devices > 08:03:48.383 alsa_dev.c ..ALSA initialized > 08:03:48.383 pjlib ..select() I/O Queue created (0x1ba09ac) > 08:03:48.390 sip_endpoint.c .Module "mod-evsub" registered > 08:03:48.390 sip_endpoint.c .Module "mod-presence" registered > 08:03:48.390 sip_endpoint.c .Module "mod-mwi" registered > 08:03:48.390 sip_endpoint.c .Module "mod-refer" registered > 08:03:48.390 sip_endpoint.c .Module "mod-pjsua-pres" registered > 08:03:48.390 sip_endpoint.c .Module "mod-pjsua-im" registered > 08:03:48.390 sip_endpoint.c .Module "mod-pjsua-options" registered > 08:03:48.391 pjsua_core.c .1 SIP worker threads created > 08:03:48.391 pjsua_core.c .pjsua version 2.6 for Linux-4.9.35/ > armv7l/glibc-2.19 initialized > 08:03:48.391 pjsua_core.c .PJSUA state changed: CREATED --> INIT > 08:03:48.391 pjsua_aud.c Setting null sound device.. > 08:03:48.391 pjsua_aud.c .Opening null sound device.. > 08:03:48.392 pjsua_core.c SIP UDP socket reachable at 192.168.178.42:5060 > 08:03:48.392 udp0x1b8a548 SIP UDP transport started, published > address is 192.168.178.42:5060 > 08:03:48.392 pjsua_core.c PJSUA state changed: INIT --> STARTING > 08:03:48.392 sip_endpoint.c .Module "mod-unsolicited-mwi" registered > 08:03:48.392 pjsua_core.c .PJSUA state changed: STARTING --> RUNNING > 08:03:48.392 pjsua_acc.c Adding account: id=sip:control@fritz.box > 08:03:48.392 pjsua_acc.c .Account sip:control@fritz.box added with id 0 > 08:03:48.392 pjsua_acc.c .Acc 0: setting registration.. > 08:03:48.394 pjsua_core.c ...TX 504 bytes Request msg REGISTER/ > cseq=12711 (tdta0x1bb43f8) 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=z9hG4bKPjgDk237BB0Z1As4Djx4OjJ6Ib5OY-hidy > Max-Forwards: 70 > From: <sip:control@fritz.box>;tag=e4CKomYR-BUoZgnDJnWct4GrGq6noRaw > To: <sip:control@fritz.box> > Call-ID: JFKgSk6EBSp.rT5Ngn2kpmBkuYfUBhMW > CSeq: 12711 REGISTER > Contact: <sip:control@192.168.178.42:5060;ob> > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, > NOTIFY, REFER, MESSAGE, OPTIONS > Content-Length: 0 > > --end msg-- > 08:03:48.394 pjsua_acc.c ..Acc 0: Registration sent > 08:03:48.402 pjsua_core.c .RX 432 bytes Response msg 401/REGISTER/ > cseq=12711 (rdata0x1b8bb7c) 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=z9hG4bKPjgDk237BB0Z1As4Djx4OjJ6Ib5OY-hidy > From: <sip:control@fritz.box>;tag=e4CKomYR-BUoZgnDJnWct4GrGq6noRaw > To: <sip:control@fritz.box>;tag=80CA5F3576C71F79 > Call-ID: JFKgSk6EBSp.rT5Ngn2kpmBkuYfUBhMW > CSeq: 12711 REGISTER > WWW-Authenticate: Digest realm="fritz.box", nonce="75A1D3FC1DE38C16" > User-Agent: FRITZ!OS > Content-Length: 0 > > --end msg-- > 08:03:48.403 pjsua_core.c ....TX 663 bytes Request msg REGISTER/ > cseq=12712 (tdta0x1bb43f8) 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=z9hG4bKPjXHEmq8A.BiZahhKpc.QOS6NYVR-.ZkTY > Max-Forwards: 70 > From: <sip:control@fritz.box>;tag=e4CKomYR-BUoZgnDJnWct4GrGq6noRaw > To: <sip:control@fritz.box> > Call-ID: JFKgSk6EBSp.rT5Ngn2kpmBkuYfUBhMW > CSeq: 12712 REGISTER > Contact: <sip:control*** Register: code= 200 > Start CALL! > MyCall::onCallState > MyCall::onCallState > MyCall::onCallMediaState > @192.168.178.42:5060;ob> > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, > NOTIFY, REFER, MESSAGE, OPTIONS > Authorization: Digest username="control", realm="fritz.box", > nonce="75A1D3FC1DE38C16", uri="sip:fritz.box", > response="d65dee7dec9d8b1a160f352bd234f602" > Content-Length: 0 > > --end msg-- > 08:03:48.410 pjsua_core.c .RX 698 bytes Response msg 200/REGISTER/ > cseq=12712 (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=z9hG4bKPjXHEmq8A.BiZahhKpc.QOS6NYVR-.ZkTY > From: <sip:control@fritz.box>;tag=e4CKomYR-BUoZgnDJnWct4GrGq6noRaw > To: <sip:control@fritz.box>;tag=7CC5E4EDC680E987 > Call-ID: JFKgSk6EBSp.rT5Ngn2kpmBkuYfUBhMW > CSeq: 12712 REGISTER > Contact: <sip:control@192.168.178.42:5060;ob>;expires=300 > 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-- > 08:03:48.410 pjsua_acc.c ....SIP outbound status for acc 0 is not active > 08:03:48.410 pjsua_acc.c ....sip:control@fritz.box: registration > success, status=200 (OK), will re-register in 300 seconds > 08:03:48.410 pjsua_acc.c ....Keep-alive timer started for acc 0, > destination:192.168.178.1:5060, interval:15s > 08:03:49.392 pjsua_aud.c Closing sound device after idle for 1 second(s) > 08:03:49.392 pjsua_aud.c .Closing null sound device.. > 08:03:58.395 pjsua_call.c !Making call with acc #0 to sip:**1@fritz.box > 08:03:58.395 pjsua_aud.c .Set sound device: capture=-99, playback=-99 > 08:03:58.395 pjsua_aud.c ..Setting null sound device.. > 08:03:58.395 pjsua_aud.c ...Opening null sound device.. > 08:03:58.395 pjsua_media.c .Call 0: initializing media.. > 08:03:58.396 pjsua_media.c ..RTP socket reachable at 192.168.178.42:4000 > 08:03:58.396 pjsua_media.c ..RTCP socket reachable at 192.168.178.42:4001 > 08:03:58.396 pjsua_media.c ..Media index 0 selected for audio call 0 > 08:03:58.399 pjsua_core.c ....TX 1071 bytes Request msg INVITE/ > cseq=5320 (tdta0x1bb8918) to UDP 192.168.178.1:5060: > INVITE sip:**1@fritz.box SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.42: > 5060;rport;branch=z9hG4bKPjoVffQxnRns2M6dRNdVofacQIBB1MU0jm > Max-Forwards: 70 > From: sip:control@fritz.box;tag=ZOKYABGChnZHwr7GpEuIJbxhc7GeLr6- > To: sip:**1@fritz.box > Contact: <sip:control@192.168.178.42:5060;ob> > Call-ID: ImFJX4NogfS.aPyhSunNEulY6K8fdePa > CSeq: 5320 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: 479 > > v=0 > o=- 3710469838 3710469838 IN IP4 192.168.178.42 > s=pjmedia > b=AS:84 > t=0 0 > a=X-nat:0 > m=audio 4000 RTP/AVP 104 98 97 99 3 0 8 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:104 iLBC/8000 > a=fmtp:104 mode=30 > a=rtpmap:98 speex/16000 > a=rtpmap:97 speex/8000 > a=rtpmap:99 speex/32000 > a=rtpmap:3 GSM/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:9 G722/8000 > a=rtpmap:96 telephone-event/8000 > a=fmtp:96 0-16 > > --end msg-- > 08:03:58.405 pjsua_core.c .RX 419 bytes Response msg 401/INVITE/ > cseq=5320 (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=z9hG4bKPjoVffQxnRns2M6dRNdVofacQIBB1MU0jm > From: <sip:control@fritz.box>;tag=ZOKYABGChnZHwr7GpEuIJbxhc7GeLr6- > To: <sip:**1@fritz.box>;tag=A7313C6763DD7B57 > Call-ID: ImFJX4NogfS.aPyhSunNEulY6K8fdePa > CSeq: 5320 INVITE > WWW-Authenticate: Digest realm="fritz.box", nonce="A7103BB430D7AE63" > User-Agent: FRITZ!OS > Content-Length: 0 > > --end msg-- > 08:03:58.405 pjsua_core.c ..TX 339 bytes Request msg ACK/cseq=5320 > (tdta0x75503bd0) to UDP 192.168.178.1:5060: > ACK sip:**1@fritz.box SIP/2.0 > Via: SIP/2.0/UD > P 192.168.178.42:5060;rport;branch=z9hG4bKPjoVffQxnRns2M6dRNdVofacQIBB1MU0jm > Max-Forwards: 70 > From: sip:control@fritz.box;tag=ZOKYABGChnZHwr7GpEuIJbxhc7GeLr6- > To: sip:**1@fritz.box;tag=A7313C6763DD7B57 > Call-ID: ImFJX4NogfS.aPyhSunNEulY6K8fdePa > CSeq: 5320 ACK > Content-Length: 0 > > --end msg-- > 08:03:58.405 pjsua_core.c .......TX 1234 bytes Request msg INVITE/ > cseq=5321 (tdta0x1bb8918) to UDP 192.168.178.1:5060: > INVITE sip:**1@fritz.box SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.42: > 5060;rport;branch=z9hG4bKPjs4xNuSbT6cqjpNaEz-LvtSgHoBqC7Ax3 > Max-Forwards: 70 > From: sip:control@fritz.box;tag=ZOKYABGChnZHwr7GpEuIJbxhc7GeLr6- > To: sip:**1@fritz.box > Contact: <sip:control@192.168.178.42:5060;ob> > Call-ID: ImFJX4NogfS.aPyhSunNEulY6K8fdePa > CSeq: 5321 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="control", realm="fritz.box", > nonce="A7103BB430D7AE63", uri="sip:**1@fritz.box", > response="1b7bf1bda6e96088bdf72820185ee781" > Content-Type: application/sdp > Content-Length: 479 > > v=0 > o=- 3710469838 3710469838 IN IP4 192.168.178.42 > s=pjmedia > b=AS:84 > t=0 0 > a=X-nat:0 > m=audio 4000 RTP/AVP 104 98 97 99 3 0 8 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:104 iLBC/8000 > a=fmtp:104 mode=30 > a=rtpmap:98 speex/16000 > a=rtpmap:97 speex/8000 > a=rtpmap:99 speex/32000 > a=rtpmap:3 GSM/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:9 G722/8000 > a=rtpmap:96 telephone-event/8000 > a=fmtp:96 0-16 > > --end msg-- > 08:03:58.425 pjsua_core.c .RX 364 bytes Response msg 100/INVITE/ > cseq=5321 (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=z9hG4bKPjs4xNuSbT6cqjpNaEz-LvtSgHoBqC7Ax3 > From: <sip:control@fritz.box>;tag=ZOKYABGChnZHwr7GpEuIJbxhc7GeLr6- > To: <sip:**1@fritz.box> > Call-ID: ImFJX4NogfS.aPyhSunNEulY6K8fdePa > CSeq: 5321 INVITE > User-Agent: AVM FRITZ!Box Fon WLAN 7390 84.06.83 (Mar 8 2017) > Content-Length: 0 > > --end msg-- > 08:03:58.458 pjsua_core.c .RX 804 bytes Response msg 183/INVITE/ > cseq=5321 (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=z9hG4bKPjs4xNuSbT6cqjpNaEz-LvtSgHoBqC7Ax3 > From: <sip:control@fritz.box>;tag=ZOKYABGChnZHwr7GpEuIJbxhc7GeLr6- > To: <sip:**1@fritz.box>;tag=F723EFB025BCF533 > Call-ID: ImFJX4NogfS.aPyhSunNEulY6K8fdePa > CSeq: 5321 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: 314 > > v=0 > o=user 15920484 15920484 IN IP4 192.168.178.1 > s=pjmedia > c=IN IP4 192.168.178.1 > t=0 0 > m=audio 7082 RTP/AVP 9 104 0 8 96 > a=rtpmap:104 iLBC/8000 > a=fmtp:104 mode=30 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:9 G722/8000 > a=rtpmap:96 telephone-event/8000 > a=fmtp:96 0-15 > a=sendrecv > a=rtcp:7083 > > --end msg-- > 08:03:58.458 pjsua_media.c .....Call 0: updating media.. > 08:03:58.458 pjsua_aud.c ......Audio channel update.. > 08:03:58.459 strm0x75507a64 .......VAD temporarily disabled > 08:03:58.459 strm0x75507a64 .......Encoder stream started > 08:03:58.459 strm0x75507a64 .......Decoder stream started > 08:03:58.459 pjsua_media.c ......Audio updated, stream #0: G722 (sendrecv) > 08:03:58.459 pjsua_aud.c .....Conf connect: 1 --> 0 > 08:03:58.459 conference.c ......Port 1 (sip:**1@fritz.box) > transmitting to port 0 (Master/sound) > 08:03:58.459 pjsua_aud.c .....Conf connect: 0 --> 1 > 08:03:58.459 conference.c ......Port 0 (Master/sound) transmitting to port 1 ( > sip:**1@fritz.box) > 08:03:58.885 stream.c G722 codec used, remote samples per frame detected = 80 > 08:03:59.095 strm0x75507a64 VAD re-enabled > 08:04:00.870 pjsua_core.c .RX 1050 bytes Response msg 200/INVITE/ > cseq=5321 (rdata0x7550169c) from UDP 192.168.178.1:5060: > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168MyCall::onCallState > MyCall::onCallState > .178.42:5060;rport=5060;branch=z9hG4bKPjs4xNuSbT6cqjpNaEz-LvtSgHoBqC7Ax3 > From: <sip:control@fritz.box>;tag=ZOKYABGChnZHwr7GpEuIJbxhc7GeLr6- > To: <sip:**1@fritz.box>;tag=F723EFB025BCF533 > Call-ID: ImFJX4NogfS.aPyhSunNEulY6K8fdePa > CSeq: 5321 INVITE > Contact: <sip:EEE303552C7E89C15FFEDA99CA2A7@192.168.178.1> > Session-Expires: 1800;refresher=uac > Min-SE: 90 > User-Agent: AVM FRITZ!Box Fon WLAN 7390 84.06.83 (Mar 8 2017) > Supported: 100rel,replaces,timer > Allow-Events: telephone-event,refer > Allow: > INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE,PUBLISH > Content-Type: application/sdp > Accept: application/sdp, multipart/mixed > Accept-Encoding: identity > Content-Length: 290 > > v=0 > o=user 15920484 15920485 IN IP4 192.168.178.1 > s=pjmedia > c=IN IP4 192.168.178.1 > t=0 0 > m=audio 7082 RTP/AVP 104 0 8 96 > a=rtpmap:104 iLBC/8000 > a=fmtp:104 mode=30 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:96 telephone-event/8000 > a=fmtp:96 0-15 > a=sendrecv > a=rtcp:7083 > > --end msg-- > 08:04:00.871 inv0x1bb445c ....SDP negotiation done, message body is ignored > 08:04:00.871 pjsua_core.c .....TX 369 bytes Request msg ACK/ > cseq=5321 (tdta0x7550cac8) to UDP 192.168.178.1:5060: > ACK sip:EEE303552C7E89C15FFEDA99CA2A7@192.168.178.1 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.42: > 5060;rport;branch=z9hG4bKPjZTvu2N7Ts4OG31LGeIs5XeeUDCmY5Iap > Max-Forwards: 70 > From: sip:control@fritz.box;tag=ZOKYABGChnZHwr7GpEuIJbxhc7GeLr6- > To: sip:**1@fritz.box;tag=F723EFB025BCF533 > Call-ID: ImFJX4NogfS.aPyhSunNEulY6K8fdePa > CSeq: 5321 ACK > Content-Length: 0 > > --end msg-- > 08:04:00.871 pjsua_call.c .Call 0 sending UPDATE for updating media > session to use only one codec > 08:04:00.872 pjsua_core.c ....TX 836 bytes Request msg UPDATE/ > cseq=5322 (tdta0x75510b58) to UDP 192.168.178.1:5060: > UPDATE sip:EEE303552C7E89C15FFEDA99CA2A7@192.168.178.1 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.42:5060;rport;branch=z9hG4bKPjOGaH3- > srAqv5CzFr.zdLzHaKV90z789N > Max-Forwards: 70 > From: sip:control@fritz.box;tag=ZOKYABGChnZHwr7GpEuIJbxhc7GeLr6- > To: sip:**1@fritz.box;tag=F723EFB025BCF533 > Contact: <sip:control@192.168.178.42:5060;ob> > Call-ID: ImFJX4NogfS.aPyhSunNEulY6K8fdePa > CSeq: 5322 UPDATE > Supported: replaces, 100rel, timer, norefersub > Session-Expires: 1800;refresher=uac > Min-SE: 90 > Content-Type: application/sdp > Content-Length: 277 > > v=0 > o=- 3710469838 3710469839 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=rtpmap:9 G722/8000 > a=rtpmap:96 telephone-event/8000 > a=fmtp:96 0-16 > a=sendrecv > > --end msg-- > 08:04:00.890 pjsua_core.c .RX 356 bytes Response msg 488/UPDATE/ > cseq=5322 (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=z9hG4bKPjOGaH3-srAqv5CzFr.zdLzHaKV90z789N > From: <sip:control@fritz.box>;tag=ZOKYABGChnZHwr7GpEuIJbxhc7GeLr6- > To: <sip:**1@fritz.box>;tag=F723EFB025BCF533 > Call-ID: ImFJX4NogfS.aPyhSunNEulY6K8fdePa > CSeq: 5322 UPDATE > User-Agent: FRITZ!OS > Content-Length: 0 > > --end msg— > > 08:04:00.951 strm0x75507a64 Bad RTP pt 104 (expecting 9) > 08:04:00.983 strm0x75507a64 Bad RTP pt 104 (expecting 9) > 08:04:01.010 strm0x75507a64 Bad RTP pt 104 (expecting 9) > 08:04:01.039 strm0x75507a64 Bad RTP pt 104 (expecting 9) > 08:04:01.072 strm0x75507a64 Bad RTP pt 104 (expecting 9) > 08:04:01.103 strm0x75507a64 Bad RTP pt 104 (expecting 9) > 08:04:01.127 strm0x75507a64 Bad RTP pt 104 (expecting 9) > 08:04:01.159 strm0x75507a64 Bad RTP pt 104 (expecting 9) > 08:04:01.192 strm0x75507a64 Bad RTP pt 104 (expecting 9) > 08:04:01.223 strm0x75507a64 Bad RTP pt 104 (expecting 9) > 08:04:01.247 strm0x75507a64 Bad RTP pt 104 (expecting 9) > 08:04:01.280 strm0x75507a64 Bad RTP pt 104 (expecting 9) > 08:04:01.311 strm0x75507a64 Bad RTP pt 104 (expecting 9) > 08:04:01.343 strm0x75507a64 Bad RTP pt 104 (expecting 9) > 08:04:01.367 strm0x75507a64 Bad RTP pt 104 (expecting 9) > 08:04:01.399 strm0x75507a64 Bad RTP pt 104 (expecting 9) > 08:04:01.431 strm0x75507a64 Bad RTP pt 104 (expecting 9) > 08:04:01.464 strm0x75507a64 Bad RTP pt 104 (expecting 9) > 08:04:01.487 strm0x75507a64 Bad RTP pt 104 (expecting 9) > 08:04:01.519 strm0x75507a64 Bad RTP pt 104 (expecting 9) > 08:04:01.551 strm0x75507a64 Bad RTP pt 104 (expecting 9) > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip@lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org