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 <sip:control@fritz.box> 08:03:48.392 pjsua_acc.c .Account sip:control@fritz.box <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: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 <sip:control@fritz.box>>;tag=e4CKomYR-BUoZgnDJnWct4GrGq6noRaw To: <sip:control@fritz.box <sip:control@fritz.box>> Call-ID: JFKgSk6EBSp.rT5Ngn2kpmBkuYfUBhMW CSeq: 12711 REGISTER Contact: <sip:control@192.168.178.42:5060;ob <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 <sip:control@fritz.box>>;tag=e4CKomYR-BUoZgnDJnWct4GrGq6noRaw To: <sip:control@fritz.box <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: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 <sip:control@fritz.box>>;tag=e4CKomYR-BUoZgnDJnWct4GrGq6noRaw To: <sip:control@fritz.box <sip:control@fritz.box>> Call-ID: JFKgSk6EBSp.rT5Ngn2kpmBkuYfUBhMW CSeq: 12712 REGISTER Contact: <sip:control*** <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 <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 <sip:control@fritz.box>>;tag=e4CKomYR-BUoZgnDJnWct4GrGq6noRaw To: <sip:control@fritz.box <sip:control@fritz.box>>;tag=7CC5E4EDC680E987 Call-ID: JFKgSk6EBSp.rT5Ngn2kpmBkuYfUBhMW CSeq: 12712 REGISTER Contact: <sip:control@192.168.178.42:5060;ob <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: <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 <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:**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- <sip:control@fritz.box;tag=ZOKYABGChnZHwr7GpEuIJbxhc7GeLr6-> To: sip:**1@fritz.box <sip:**1@fritz.box> Contact: <sip:control@192.168.178.42:5060;ob <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 <sip:control@fritz.box>>;tag=ZOKYABGChnZHwr7GpEuIJbxhc7GeLr6- To: <sip:**1@fritz.box <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:**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- <sip:control@fritz.box;tag=ZOKYABGChnZHwr7GpEuIJbxhc7GeLr6-> To: sip:**1@fritz.box;tag=A7313C6763DD7B57 <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:**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- <sip:control@fritz.box;tag=ZOKYABGChnZHwr7GpEuIJbxhc7GeLr6-> To: sip:**1@fritz.box <sip:**1@fritz.box> Contact: <sip:control@192.168.178.42:5060;ob <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 <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 <sip:control@fritz.box>>;tag=ZOKYABGChnZHwr7GpEuIJbxhc7GeLr6- To: <sip:**1@fritz.box <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 <sip:control@fritz.box>>;tag=ZOKYABGChnZHwr7GpEuIJbxhc7GeLr6- To: <sip:**1@fritz.box <sip:**1@fritz.box>>;tag=F723EFB025BCF533 Call-ID: ImFJX4NogfS.aPyhSunNEulY6K8fdePa CSeq: 5321 INVITE Contact: <sip:EEE303552C7E89C15FFEDA99CA2A7@192.168.178.1 <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 <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 <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 <sip:control@fritz.box>>;tag=ZOKYABGChnZHwr7GpEuIJbxhc7GeLr6- To: <sip:**1@fritz.box <sip:**1@fritz.box>>;tag=F723EFB025BCF533 Call-ID: ImFJX4NogfS.aPyhSunNEulY6K8fdePa CSeq: 5321 INVITE Contact: <sip:EEE303552C7E89C15FFEDA99CA2A7@192.168.178.1 <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: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- <sip:control@fritz.box;tag=ZOKYABGChnZHwr7GpEuIJbxhc7GeLr6-> To: sip:**1@fritz.box;tag=F723EFB025BCF533 <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: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- <sip:control@fritz.box;tag=ZOKYABGChnZHwr7GpEuIJbxhc7GeLr6-> To: sip:**1@fritz.box;tag=F723EFB025BCF533 <sip:**1@fritz.box;tag=F723EFB025BCF533> Contact: <sip:control@192.168.178.42:5060;ob <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 <sip:control@fritz.box>>;tag=ZOKYABGChnZHwr7GpEuIJbxhc7GeLr6- To: <sip:**1@fritz.box <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)

On 08/02/2017 05:34 PM, Kevin Rombach via pjsip wrote: > 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: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- <sip:control@fritz.box;tag=ZOKYABGChnZHwr7GpEuIJbxhc7GeLr6-> > To: sip:**1@fritz.box;tag=F723EFB025BCF533 <sip:**1@fritz.box;tag=F723EFB025BCF533> > Contact: <sip:control@192.168.178.42:5060;ob <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 <sip:control@fritz.box>>;tag=ZOKYABGChnZHwr7GpEuIJbxhc7GeLr6- > To: <sip:**1@fritz.box <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) > The answer is here: 08:04:00.871 pjsua_call.c .Call 0 sending UPDATE for updating media session to use only one codec PJ is trying to change the session to use only one codec after it has been established. However, the Fritzbox rejects the UPDATE (which _should_ be fine) and the session continues as is. IIRC PJ cannot deal with multiple codecs inside a single session, which is why it tries to change the session to use only one codec. It would be interesting to know why the Fritzbox rejects the UPDATE, maybe there are some logs on the Fritzbox? A (cheap) solution would be to limit the codecs that either side supports/offers; (i.e. only allow 'g722' in the fritzbox or PJ). Regards, Andreas

Bad RTP pt 0 (expecting 9)

Bad RTP pt 104 (expecting 9)

Thanks i solved it by disabling all other codes then “G722” in my PJLib. But i think this is not the best solution? :-/ I also tried to just disable only the iLBC codec which is requested from the fritz.box. But when i do this im getting Bad RTP pt 0 (expecting 9) instead of >> Bad RTP pt 104 (expecting 9) Any idea? > Am 03.08.2017 um 06:35 schrieb Andreas Wehrmann <a.wehrmann@yandex.com>: > > On 08/02/2017 05:34 PM, Kevin Rombach via pjsip wrote: >> 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:EEE303552C7E89C15FFEDA99CA2A7@192.168.178.1> <sip:EEE303552C7E89C15FFEDA99CA2A7@192.168.178.1 <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- <sip:control@fritz.box;tag=ZOKYABGChnZHwr7GpEuIJbxhc7GeLr6-><sip:control@fritz.box;tag=ZOKYABGChnZHwr7GpEuIJbxhc7GeLr6- <sip:control@fritz.box;tag=ZOKYABGChnZHwr7GpEuIJbxhc7GeLr6->> >> To: sip:**1@fritz.box;tag=F723EFB025BCF533 <sip:**1@fritz.box;tag=F723EFB025BCF533> <sip:**1@fritz.box;tag=F723EFB025BCF533 <sip:**1@fritz.box;tag=F723EFB025BCF533>> >> Contact: <sip:control@192.168.178.42:5060;ob <sip:control@192.168.178.42:5060;ob> <sip:control@192.168.178.42:5060;ob <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 <sip:control@fritz.box> <sip:control@fritz.box <sip:control@fritz.box>>>;tag=ZOKYABGChnZHwr7GpEuIJbxhc7GeLr6- >> To: <sip:**1@fritz.box <sip:**1@fritz.box> <sip:**1@fritz.box <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) >> > > The answer is here: > > 08:04:00.871 pjsua_call.c .Call 0 sending UPDATE for updating media session to use only one codec > > > PJ is trying to change the session to use only one codec after it has been established. > However, the Fritzbox rejects the UPDATE (which _should_ be fine) and the session continues as is. > > IIRC PJ cannot deal with multiple codecs inside a single session, > which is why it tries to change the session to use only one codec. > > It would be interesting to know why the Fritzbox rejects the UPDATE, maybe there are some logs on the Fritzbox? > > A (cheap) solution would be to limit the codecs that either side supports/offers; (i.e. only allow 'g722' in the fritzbox or PJ). > > > Regards, > Andreas > > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org <http://blog.pjsip.org/> > > pjsip mailing list > pjsip@lists.pjsip.org <mailto:pjsip@lists.pjsip.org> > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org <http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org>

Bad RTP pt 0 (expecting 9)

Bad RTP pt 104 (expecting 9)

On 08/03/2017 07:47 PM, Kevin Rombach via pjsip wrote: > Thanks i solved it by disabling all other codes then “G722” in my PJLib. But i think this is not the best solution? :-/ > > I also tried to just disable only the iLBC codec which is requested from the fritz.box. But when i do this im getting > > Bad RTP pt 0 (expecting 9) > > instead of > >>> Bad RTP pt 104 (expecting 9) > Any idea? > Not really; I guess the same sequence of events is happening there as well. Did you really _disable_ all other codecs, so that they don't appear inside the SDP message, or did you adjust codec priorities to change the order in which they appear inside the SDP message? Whatever the case, I don't see why the Fritzbox is rejecting the UPDATE... Is this a known limitation of theirs, maybe? Regards

Bad RTP pt 0 (expecting 9)

Bad RTP pt 104 (expecting 9)

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? > Whatever the case, I don't see why the Fritzbox is rejecting the UPDATE... > Is this a known limitation of theirs, maybe? Im very new in this VOIP stuff so i cant say anything about it :/ But i searched in the whole webinterface of my fritzbox for any VOIP codec setting but i could not find anything. Thats why i now disabled all but 1 coded in my pjlib. > Am 04.08.2017 um 06:34 schrieb Andreas Wehrmann <a.wehrmann@yandex.com>: > > On 08/03/2017 07:47 PM, Kevin Rombach via pjsip wrote: >> Thanks i solved it by disabling all other codes then “G722” in my PJLib. But i think this is not the best solution? :-/ >> >> I also tried to just disable only the iLBC codec which is requested from the fritz.box. But when i do this im getting >> >> Bad RTP pt 0 (expecting 9) >> >> instead of >> >>>> Bad RTP pt 104 (expecting 9) >> Any idea? >> > > Not really; I guess the same sequence of events is happening there as well. > Did you really _disable_ all other codecs, so that they don't appear inside the SDP message, > or did you adjust codec priorities to change the order in which they appear inside the SDP message? > > Whatever the case, I don't see why the Fritzbox is rejecting the UPDATE... > Is this a known limitation of theirs, maybe? > > Regards > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip@lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org

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.

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: [pjsip] 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. 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@fritz.box 06:59:57.999 pjsua_acc.c .Account sip:doorz-control@fritz.box 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@fritz.box>;tag=ENVAypb9UtGAaaD3W8FKNqWiSSKFEYHg To: <sip:doorz-control@fritz.box> Call-ID: VqR3bDyVbnW.Nu8F6SCeFy18wv5NHHSO CSeq: 61758 REGISTER Contact: <sip:doorz-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-- 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@fritz.box>;tag=ENVAypb9UtGAaaD3W8FKNqWiSSKFEYHg To: <sip:doorz-control@fritz.box>;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@fritz.box>;tag=ENVAypb9UtGAaa*** Register: code= 200 Start CALL! MyCall::onCallState To: <sip:doorz-control@fritz.box> Call-ID: VqR3bDyVbnW.Nu8F6SCeFy18wv5NHHSO CSeq: 61759 REGISTER Contact: <sip:doorz-control@192.168.178.42:5060;ob> 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@fritz.box>;tag=ENVAypb9UtGAaaD3W8FKNqWiSSKFEYHg To: <sip:doorz-control@fritz.box>;tag=22DE2A38BEF81644 Call-ID: VqR3bDyVbnW.Nu8F6SCeFy18wv5NHHSO CSeq: 61759 REGISTER Contact: <sip:doorz-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-- 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@fritz.box: 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@fritz.box 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@fritz.box SIP/2.0 Via: SIP/2.0/UDP 192.168.178.42:5060;rport;branch=z9hG4bKPjfXj0f0BTz7AIM-YUaweJGxglRroSvjsf Max-Forwards: 70 From: sip:doorz-control@fritz.box;tag=JtPN-VUUOmhq3.I4c4KkZcjZu-xz.95D To: sip:**1@fritz.box Contact: <sip:doorz-control@192.168.178.42:5060;ob> 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@fritz.box>;tag=JtPN-VUUOmhq3.I4c4KkZcjZu-xz.95D To: <sip:**1@fritz.box>;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@fritz.box SIP/2.0 Via: SIP/2.0/UDP 192.168.178.42:5060;rport;branch=z9hG4bKPjfXj0f0BMyCall::onCallState MyCall::onCallMediaState Max-Forwards: 70 From: sip:doorz-control@fritz.box;tag=JtPN-VUUOmhq3.I4c4KkZcjZu-xz.95D To: sip:**1@fritz.box;tag=AAEAB2A16C04D12D 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@fritz.box SIP/2.0 Via: SIP/2.0/UDP 192.168.178.42:5060;rport;branch=z9hG4bKPjzuUpsmNa3R2odjt52f-Zn0ogRbiQ0auT Max-Forwards: 70 From: sip:doorz-control@fritz.box;tag=JtPN-VUUOmhq3.I4c4KkZcjZu-xz.95D To: sip:**1@fritz.box Contact: <sip:doorz-control@192.168.178.42:5060;ob> 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@fritz.box", 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@fritz.box>;tag=JtPN-VUUOmhq3.I4c4KkZcjZu-xz.95D To: <sip:**1@fritz.box> 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@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 > Am 04.08.2017 um 06:55 schrieb Andreas Wehrmann <a.wehrmann@yandex.com>: > > 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. > > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip@lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org

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: > > [pjsip] 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@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...

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? > Am 04.08.2017 um 07:20 schrieb Andreas Wehrmann <a.wehrmann@yandex.com>: > > 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: >> >> [pjsip] 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@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

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... >> >> >> >>