I recently setup a FreePBX for a business wanting to allow some of their users to work remotely. I used a Grandstream GXW410X gateway and a virtualized FreePBX 15.0.16.44 (16.9.0)
All hardware phones are Grandstream GXP2170 and soft phones are MicroSIP.
95% of calls are working fine but the customer is reporting random call dropping to/from some of the soft phones. In order to access the business application, the remote users are connected via VPN.
Looking through the call logs, I can’t seem to find any problems other than MicroSIP reporting occasionally high jitter. There does not appear to be a correlation between high jitter and dropped calls. The customer has 300 down / 15 up and if I can prove the problem is bandwidth related would be willing to upgrade.
The user at 1103 called the hardphone at 106 which call forwarded to the softphone at 1106. Other than the jitter nothing appears out of the ordinary to me. I am using chan_sip cause I couldn’t get the PSTN gateway to work with pjsip (plus I’m just more comfortable with it).
Below is part of the MicroSIP log. I will have to split it into two posts.
13:43:34.054 sip_endpoint.c Processing incoming message: Request msg INVITE/cseq=102 (rdata07D5AEEC)
13:43:34.054 pjsua_core.c .RX 1058 bytes Request msg INVITE/cseq=102 (rdata07D5AEEC) from UDP 10.99.11.21:5160:
INVITE sip:[email protected]:55374;ob SIP/2.0
Via: SIP/2.0/UDP 10.99.11.21:5160;branch=z9hG4bK38147f45
Max-Forwards: 70
From: "Virtual Olfat" <sip:[email protected]:5160>;tag=as4ef92226
To: <sip:[email protected]:55374;ob>
Contact: <sip:[email protected]:5160>
Call-ID: [email protected]:5160
CSeq: 102 INVITE
User-Agent: FPBX-15.0.16.44(16.9.0)
Date: Mon, 27 Apr 2020 20:43:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
P-Asserted-Identity: "Virtual Olfat" <sip:[email protected]>
Diversion: <sip:[email protected]>;reason=unconditional
Content-Type: application/sdp
Content-Length: 351
v=0
o=root 1671105369 1671105369 IN IP4 10.99.11.21
s=Asterisk PBX 16.9.0
c=IN IP4 10.99.11.21
t=0 0
m=audio 19378 RTP/AVP 0 8 3 111 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
--end msg--
13:43:34.054 pjsua_call.c .Incoming Request msg INVITE/cseq=102 (rdata07D5AEEC)
13:43:34.054 tsx0732D9B4 ...Transaction created for Request msg INVITE/cseq=102 (rdata07D5AEEC)
13:43:34.054 tsx0732D9B4 ..Incoming Request msg INVITE/cseq=102 (rdata07D5AEEC) in state Null
13:43:34.055 tsx0732D9B4 ...State changed from Null to Trying, event=RX_MSG
13:43:34.055 dlg0733320C ....Transaction tsx0732D9B4 state changed to Trying
13:43:34.055 dlg0733320C ..UAS dialog created
13:43:34.055 dlg0733320C ..Module mod-invite added as dialog usage, data=072D882C
13:43:34.055 dlg0733320C ...Session count inc to 3 by mod-invite
13:43:34.055 inv0733320C ..UAS invite session created for dialog dlg0733320C
13:43:34.055 dlg0733320C ...Session count inc to 3 by mod-pjsua
13:43:34.055 pjsua_media.c ..Call 3: initializing media..
13:43:34.074 pjsua_media.c ...RTP socket reachable at 10.99.11.122:4000
13:43:34.074 pjsua_media.c ...RTCP socket reachable at 10.99.11.122:4001
13:43:34.074 srtp07D6C348 ...SRTP keying SDES created
13:43:34.074 pjsua_media.c ...Media index 0 selected for audio call 3
13:43:34.074 pjsua_media.c ...Call 3: media transport initialization complete: Success
13:43:34.074 dlg0733320C ...Session count dec to 3 by mod-pjsua
13:43:34.074 pjsua_call.c ..Call 3: remote NAT type is 0 (Unknown)
13:43:34.074 endpoint ...Response msg 100/INVITE/cseq=102 (tdta073429E4) created
13:43:34.074 dlg0733320C ...Initial answer Response msg 100/INVITE/cseq=102 (tdta073429E4)
13:43:34.074 inv0733320C ...Sending Response msg 100/INVITE/cseq=102 (tdta073429E4)
13:43:34.074 dlg0733320C ....Sending Response msg 100/INVITE/cseq=102 (tdta073429E4)
13:43:34.074 tsx0732D9B4 ....Sending Response msg 100/INVITE/cseq=102 (tdta073429E4) in state Trying
13:43:34.074 sip_resolve.c .....Target '10.99.11.21:5160' type=UDP resolved to '10.99.11.21:5160' type=UDP (UDP transport)
13:43:34.074 pjsua_core.c .....TX 292 bytes Response msg 100/INVITE/cseq=102 (tdta073429E4) to UDP 10.99.11.21:5160:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.99.11.21:5160;received=10.99.11.21;branch=z9hG4bK38147f45
Call-ID: [email protected]:5160
From: "Virtual Olfat" <sip:[email protected]>;tag=as4ef92226
To: <sip:[email protected];ob>
CSeq: 102 INVITE
Content-Length: 0
--end msg--
13:43:34.075 tsx0732D9B4 .....State changed from Trying to Proceeding, event=TX_MSG
13:43:34.075 dlg0733320C ......Transaction tsx0732D9B4 state changed to Proceeding
13:43:34.075 pjsua_call.c ..Answering call 3: code=180
13:43:34.075 sip_transport. ....Tx data Response msg 100/INVITE/cseq=102 (tdta073419DC) cloned
13:43:34.075 inv0733320C ....Sending Response msg 180/INVITE/cseq=102 (tdta073419DC)
13:43:34.075 dlg0733320C .....Sending Response msg 180/INVITE/cseq=102 (tdta073419DC)
13:43:34.075 tsx0732D9B4 .....Sending Response msg 180/INVITE/cseq=102 (tdta073419DC) in state Proceeding
13:43:34.075 tdta073429E4 ......Destroying txdata Response msg 100/INVITE/cseq=102 (tdta073429E4)
13:43:34.075 pjsua_core.c ......TX 478 bytes Response msg 180/INVITE/cseq=102 (tdta073419DC) to UDP 10.99.11.21:5160:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.99.11.21:5160;received=10.99.11.21;branch=z9hG4bK38147f45
Call-ID: [email protected]:5160
From: "Virtual Olfat" <sip:[email protected]>;tag=as4ef92226
To: <sip:[email protected];ob>;tag=e7c5ac89fa494c88a3a1733820865020
CSeq: 102 INVITE
Contact: "Anna" <sip:[email protected]:55374;ob>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length: 0
--end msg--
13:43:34.075 tsx0732D9B4 ......State changed from Proceeding to Proceeding, event=TX_MSG
13:43:34.075 dlg0733320C .......Transaction tsx0732D9B4 state changed to Proceeding
13:43:34.762 pjsua_aud.c !Creating file player: ringtone.wav..
13:43:34.770 wav_player.c .File player 'ringtone.wav' created: samp.rate=8000, ch=1, bufsize=4KB, filesize=48KB
13:43:34.770 pjsua_aud.c .Player created, id=0, slot=1
13:43:34.771 pjsua_aud.c Conf connect: 1 --> 0
13:43:34.771 pjsua_aud.c .Set sound device: capture=-1, playback=-2
13:43:34.771 pjsua_aud.c ..Opening sound device (speaker + mic) PCM@8000/1/20ms
13:43:34.822 wmme_dev.c ... WaveAPI Sound player "Wave mapper" initialized (format=PCM, clock_rate=8000, channel_count=1, samples_per_frame=160 (20ms))
13:43:34.835 wmme_dev.c ... WaveAPI Sound recorder "Wave mapper" initialized (format=PCM, clock_rate=8000, channel_count=1, samples_per_frame=160 (20ms))
13:43:34.836 ec07D59840 ...Creating WebRTC AEC
13:43:34.837 echo_webrtc.c ...WebRTC AEC successfully created with options 0
13:43:34.837 ec07D59840 ...Using delay buffer with WSOLA.
13:43:34.837 ec07D59840 ...WebRTC AEC created, clock_rate=8000, channel=1, samples per frame=160, tail length=200 ms, latency=100 ms
13:43:34.837 wmme_dev.c ...WMME playback stream started
13:43:34.837 wmme_dev.c ...WMME capture stream started
13:43:34.838 conference.c .Port 1 (ringtone.wav) transmitting to port 0 (Wave mapper)
13:43:34.873 ec07D59840 !Prefetching..
13:43:34.893 ec07D59840 Prefetching..
13:43:34.914 ec07D59840 Prefetching..
13:43:34.933 ec07D59840 Prefetching..
13:43:34.953 ec07D59840 Prefetching..
13:43:34.973 ec07D59840 Prefetching..
13:43:34.975 ec07D59840 Latency bufferring complete
13:43:37.875 wav_player.c File port ringtone.wav EOF
13:43:37.875 wav_player.c File port ringtone.wav rewinding..
13:43:38.127 pjsua_acc.c !Sending 2 bytes keep-alive packet for acc 0 to 10.99.11.21:5160
13:43:38.127 tdta07346A04 Destroying txdata raw
13:43:40.875 wav_player.c !File port ringtone.wav EOF
13:43:40.875 wav_player.c File port ringtone.wav rewinding..
13:43:43.875 wav_player.c File port ringtone.wav EOF
13:43:43.875 wav_player.c File port ringtone.wav rewinding..
13:43:46.874 wav_player.c File port ringtone.wav EOF
13:43:46.874 wav_player.c File port ringtone.wav rewinding..
13:43:47.824 pjsua_call.c !Answering call 3: code=200
13:43:47.824 sip_transport. ..Tx data Response msg 180/INVITE/cseq=102 (tdta07346A04) cloned
13:43:47.824 inv0733320C ..SDP negotiation done: Success
13:43:47.824 pjsua_call.c ...Call 3: remote NAT type is 0 (Unknown)
13:43:47.824 pjsua_media.c ...Call 3: updating media..
13:43:47.824 pjsua_media.c .....Media stream call03:0 is destroyed
13:43:47.824 pjsua_aud.c ....Audio channel update..
13:43:47.824 rtp.c .....pjmedia_rtp_session_init: ses=0730E3DC, default_pt=0, ssrc=0x43654e38
13:43:47.824 rtp.c .....pjmedia_rtp_session_init: ses=0730EA64, default_pt=0, ssrc=0x43654e38
13:43:47.824 udp0731A008 .....SO_RCVBUF set to 65536
13:43:47.824 udp0731A008 .....SO_SNDBUF set to 65536
13:43:47.824 stream.c .....Stream strm0737E12C created
13:43:47.824 strm0737E12C .....Encoder stream started
13:43:47.824 strm0737E12C .....Decoder stream started
13:43:47.824 pjsua_media.c ....Audio updated, stream #0: PCMU (sendrecv)
13:43:47.824 pjsua_aud.c ...Conf connect: 2 --> 0
13:43:47.824 conference.c ....Port 2 (sip:[email protected]:5160) transmitting to port 0 (Wave mapper)
13:43:47.825 pjsua_aud.c ...Conf connect: 0 --> 2
13:43:47.825 conference.c ....Port 0 (Wave mapper) transmitting to port 2 (sip:[email protected]:5160)
13:43:47.825 inv0733320C ..Sending Response msg 200/INVITE/cseq=102 (tdta07346A04)
13:43:47.825 dlg0733320C ...Sending Response msg 200/INVITE/cseq=102 (tdta07346A04)
13:43:47.825 tsx0732D9B4 ...Sending Response msg 200/INVITE/cseq=102 (tdta07346A04) in state Proceeding
13:43:47.825 tdta073419DC ....Destroying txdata Response msg 180/INVITE/cseq=102 (tdta073419DC)
13:43:47.825 pjsua_core.c ....TX 871 bytes Response msg 200/INVITE/cseq=102 (tdta07346A04) to UDP 10.99.11.21:5160:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.99.11.21:5160;received=10.99.11.21;branch=z9hG4bK38147f45
Call-ID: [email protected]:5160
From: "Virtual Olfat" <sip:[email protected]>;tag=as4ef92226
To: <sip:[email protected];ob>;tag=e7c5ac89fa494c88a3a1733820865020
CSeq: 102 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Contact: "Anna" <sip:[email protected]:55374;ob>
Supported: replaces, 100rel, timer, norefersub
Content-Type: application/sdp
Content-Length: 316
v=0
o=- 3796983814 3796983815 IN IP4 10.99.11.122
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4000 RTP/AVP 0 101
c=IN IP4 10.99.11.122
b=TIAS:64000
a=rtcp:4001 IN IP4 10.99.11.122
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:1130712632 cname:456d7e0e06e30a6c
--end msg--
13:43:47.825 tsx0732D9B4 ....State changed from Proceeding to Completed, event=TX_MSG
13:43:47.825 dlg0733320C .....Transaction tsx0732D9B4 state changed to Completed
13:43:47.825 pjsua_aud.c Conf disconnect: 1 -x- 0
13:43:47.825 conference.c .Port 1 (ringtone.wav) stop transmitting to port 0 (Wave mapper)
13:43:47.825 pjsua_aud.c Destroying player 0..
13:43:47.834 strm0737E12C !Jitter buffer is bufferring (prefetch=0)
13:43:47.834 strm0737E12C Start talksprut..
13:43:48.221 strm0737E12C !RTP status: badpt=0, badssrc=0, dup=0, outorder=0, probation=-1, restart=0
13:43:48.234 strm0737E12C !Jitter buffer starts returning normal frames (after 20 empty/lost)
13:43:48.324 tsx0732D9B4 !Retransmit timer event
13:43:48.324 tsx0732D9B4 .Retransmiting Response msg 200/INVITE/cseq=102 (tdta07346A04), count=0, restart?=1
13:43:48.324 pjsua_core.c .TX 871 bytes Response msg 200/INVITE/cseq=102 (tdta07346A04) to UDP 10.99.11.21:5160:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.99.11.21:5160;received=10.99.11.21;branch=z9hG4bK38147f45
Call-ID: [email protected]:5160
From: "Virtual Olfat" <sip:[email protected]>;tag=as4ef92226
To: <sip:[email protected];ob>;tag=e7c5ac89fa494c88a3a1733820865020
CSeq: 102 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Contact: "Anna" <sip:[email protected]:55374;ob>
Supported: replaces, 100rel, timer, norefersub
Content-Type: application/sdp
Content-Length: 316
v=0
o=- 3796983814 3796983815 IN IP4 10.99.11.122
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4000 RTP/AVP 0 101
c=IN IP4 10.99.11.122
b=TIAS:64000
a=rtcp:4001 IN IP4 10.99.11.122
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:1130712632 cname:456d7e0e06e30a6c
--end msg--
13:43:48.854 strm0737E12C !Frame lost, recovered!
13:43:48.855 strm0737E12C Jitter buffer starts returning normal frames (after 1 empty/lost)
13:43:49.034 strm0737E12C Frame lost, recovered!
13:43:49.055 strm0737E12C Jitter buffer starts returning normal frames (after 1 empty/lost)
13:43:49.234 strm0737E12C Frame lost, recovered!
13:43:49.234 strm0737E12C Jitter buffer starts returning normal frames (after 1 empty/lost)
13:43:49.326 tsx0732D9B4 !Retransmit timer event
13:43:49.326 tsx0732D9B4 .Retransmiting Response msg 200/INVITE/cseq=102 (tdta07346A04), count=1, restart?=1
13:43:49.326 pjsua_core.c .TX 871 bytes Response msg 200/INVITE/cseq=102 (tdta07346A04) to UDP 10.99.11.21:5160:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.99.11.21:5160;received=10.99.11.21;branch=z9hG4bK38147f45
Call-ID: [email protected]:5160
From: "Virtual Olfat" <sip:[email protected]>;tag=as4ef92226
To: <sip:[email protected];ob>;tag=e7c5ac89fa494c88a3a1733820865020
CSeq: 102 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Contact: "Anna" <sip:[email protected]:55374;ob>
Supported: replaces, 100rel, timer, norefersub
Content-Type: application/sdp
Content-Length: 316
v=0
o=- 3796983814 3796983815 IN IP4 10.99.11.122
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4000 RTP/AVP 0 101
c=IN IP4 10.99.11.122
b=TIAS:64000
a=rtcp:4001 IN IP4 10.99.11.122
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:1130712632 cname:456d7e0e06e30a6c
--end msg--
13:43:49.415 strm0737E12C !Frame lost, recovered!
13:43:49.434 strm0737E12C Jitter buffer starts returning normal frames (after 1 empty/lost)
13:43:49.615 strm0737E12C Frame lost, recovered!
13:43:49.615 strm0737E12C Jitter buffer starts returning normal frames (after 1 empty/lost)
13:43:49.795 strm0737E12C Frame lost, recovered!
13:43:49.814 strm0737E12C Jitter buffer starts returning normal frames (after 1 empty/lost)
13:43:49.995 strm0737E12C Frame lost, recovered!
13:43:49.995 strm0737E12C Jitter buffer starts returning normal frames (after 1 empty/lost)
13:43:50.174 strm0737E12C Frame lost, recovered!
13:43:50.195 strm0737E12C Jitter buffer starts returning normal frames (after 1 empty/lost)
13:43:50.375 strm0737E12C Frame lost, recovered!
13:43:50.375 strm0737E12C Jitter buffer starts returning normal frames (after 1 empty/lost)
13:43:50.555 strm0737E12C Frame lost, recovered!
13:43:50.574 strm0737E12C Jitter buffer starts returning normal frames (after 1 empty/lost)
13:43:50.754 strm0737E12C Frame lost, recovered!
13:43:50.754 strm0737E12C Jitter buffer starts returning normal frames (after 1 empty/lost)
13:43:50.935 strm0737E12C Frame lost, recovered!
13:43:50.954 strm0737E12C Jitter buffer starts returning normal frames (after 1 empty/lost)
13:43:51.134 strm0737E12C Frame lost, recovered!
13:43:51.135 strm0737E12C Jitter buffer starts returning normal frames (after 1 empty/lost)
13:43:51.314 strm0737E12C Frame lost, recovered!
13:43:51.326 tsx0732D9B4 !Retransmit timer event
13:43:51.326 tsx0732D9B4 .Retransmiting Response msg 200/INVITE/cseq=102 (tdta07346A04), count=2, restart?=1
13:43:51.326 pjsua_core.c .TX 871 bytes Response msg 200/INVITE/cseq=102 (tdta07346A04) to UDP 10.99.11.21:5160:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.99.11.21:5160;received=10.99.11.21;branch=z9hG4bK38147f45
Call-ID: [email protected]:5160
From: "Virtual Olfat" <sip:[email protected]>;tag=as4ef92226
To: <sip:[email protected];ob>;tag=e7c5ac89fa494c88a3a1733820865020
CSeq: 102 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Contact: "Anna" <sip:[email protected]:55374;ob>
Supported: replaces, 100rel, timer, norefersub
Content-Type: application/sdp
Content-Length: 316
v=0
o=- 3796983814 3796983815 IN IP4 10.99.11.122
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4000 RTP/AVP 0 101
c=IN IP4 10.99.11.122
b=TIAS:64000
a=rtcp:4001 IN IP4 10.99.11.122
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:1130712632 cname:456d7e0e06e30a6c
--end msg--
13:43:51.335 strm0737E12C !Jitter buffer starts returning normal frames (after 1 empty/lost)
13:43:51.515 strm0737E12C Frame lost, recovered!
13:43:51.515 strm0737E12C Jitter buffer starts returning normal frames (after 1 empty/lost)
13:43:51.695 strm0737E12C Frame lost, recovered!
13:43:51.714 strm0737E12C Jitter buffer starts returning normal frames (after 1 empty/lost)
13:43:51.895 strm0737E12C Frame lost, recovered!
13:43:51.915 strm0737E12C Jitter buffer starts returning normal frames (after 1 empty/lost)
13:43:52.095 strm0737E12C Frame lost, recovered!
13:43:52.115 strm0737E12C Jitter buffer starts returning normal frames (after 1 empty/lost)
13:43:52.295 strm0737E12C Frame lost, recovered!
13:43:52.315 strm0737E12C Jitter buffer starts returning normal frames (after 1 empty/lost)
13:43:52.494 strm0737E12C Frame lost, recovered!
13:43:52.494 strm0737E12C Jitter buffer starts returning normal frames (after 1 empty/lost)
13:43:52.694 strm0737E12C Frame lost, recovered!
13:43:52.714 strm0737E12C Jitter buffer starts returning normal frames (after 1 empty/lost)
13:43:53.034 strm0737E12C Frame lost, recovered!
13:43:53.034 strm0737E12C Jitter buffer starts returning normal frames (after 1 empty/lost)
13:43:53.128 pjsua_acc.c !Sending 2 bytes keep-alive packet for acc 0 to 10.99.11.21:5160
13:43:53.128 tdta07348A14 Destroying txdata raw
13:43:53.208 sip_endpoint.c Processing incoming message: Request msg OPTIONS/cseq=102 (rdata07D5AEEC)
13:43:53.208 pjsua_core.c .RX 560 bytes Request msg OPTIONS/cseq=102 (rdata07D5AEEC) from UDP 10.99.11.21:5160:
OPTIONS sip:[email protected]:55374;ob SIP/2.0
Via: SIP/2.0/UDP 10.99.11.21:5160;branch=z9hG4bK797f2066
Max-Forwards: 70
From: "Unknown" <sip:[email protected]:5160>;tag=as51fca75b
To: <sip:[email protected]:55374;ob>
Contact: <sip:[email protected]:5160>
Call-ID: [email protected]:5160
CSeq: 102 OPTIONS
User-Agent: FPBX-15.0.16.44(16.9.0)
Date: Mon, 27 Apr 2020 20:43:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
--end msg--
13:43:53.208 endpoint .Response msg 200/OPTIONS/cseq=102 (tdta07348A14) created
13:43:53.208 sip_resolve.c .Target '10.99.11.21:5160' type=UDP resolved to '10.99.11.21:5160' type=UDP (UDP transport)
13:43:53.208 pjsua_core.c .TX 709 bytes Response msg 200/OPTIONS/cseq=102 (tdta07348A14) to UDP 10.99.11.21:5160:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.99.11.21:5160;received=10.99.11.21;branch=z9hG4bK797f2066
Call-ID: [email protected]:5160
From: "Unknown" <sip:[email protected]>;tag=as51fca75b
To: <sip:[email protected];ob>;tag=z9hG4bK797f2066
CSeq: 102 OPTIONS
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain
Supported: replaces, 100rel, timer, norefersub
Allow-Events: presence, message-summary, refer
User-Agent: MicroSIP/3.19.28
Content-Length: 0
--end msg--
13:43:53.208 tdta07348A14 .Destroying txdata Response msg 200/OPTIONS/cseq=102 (tdta07348A14)
13:43:53.355 strm0737E12C !Frame lost, recovered!
13:43:53.355 strm0737E12C Jitter buffer starts returning normal frames (after 1 empty/lost)
13:43:54.195 strm0737E12C Jitter buffer empty (prefetch=0), plc invoked
13:43:54.375 strm0737E12C Jitter buffer starts returning normal frames (after 9 empty/lost)
13:43:54.414 strm0737E12C Jitter buffer empty (prefetch=0), plc invoked
13:43:54.435 strm0737E12C Jitter buffer starts returning normal frames (after 1 empty/lost)
13:43:54.755 strm0737E12C Frame lost, recovered!