[Live-devel] PlaySIP SegFault
NADEAU Frederic -EXT
frederic.nadeau at transport.alstom.com
Fri Oct 12 04:15:21 PDT 2012
There is another error in playsip.
To fix that issue, we used :
# playSIP -u nouser nopass 10.0.0.1 sip :100 at 10.0.0.1, I didn't look at the 2012-10-11 fix, but from what I know the issue is with the authenticator.
But there is more. Latest code(as of 2012-10-11) gives this :
(sorry for the long post, comments continue in middle and at the end)
################################################################################################
fnadeau at ip-10-xxx-xxx-xxx:~/live/testProgs> ./playSIP -Q sip:8355 at ideasip.com
Sending request: INVITE sip:8355 at ideasip.com SIP/2.0
From: 8355 <sip:8355 at 10.6.161.115>;tag=433402263
Via: SIP/2.0/UDP 10.6.161.115:55386
To: sip:8355 at ideasip.com
Contact: sip:8355 at 10.6.161.115:55386
Call-ID: 984943604 at 10.6.161.115
CSeq: 1 INVITE
Content-Type: application/sdp
User-Agent: playSIP (LIVE555 Streaming Media v2012.10.11)
Content-Length: 115
v=0
o=- 984943604 0 IN IP4 10.6.161.115
s=playSIP session
c=IN IP4 10.6.161.115
t=0 0
m=audio 8000 RTP/AVP 0
Received INVITE response: SIP/2.0 100 trying -- your call is important to us
From: 8355 <sip:8355 at 10.6.161.115>;tag=433402263
Via: SIP/2.0/UDP 10.6.161.115:55386;received=50.16.169.79
To: sip:8355 at ideasip.com
Call-ID: 984943604 at 10.6.161.115
CSeq: 1 INVITE
Server: Sip EXpress router (0.9.6 (i386/linux))
Content-Length: 0
Warning: 392 208.97.25.11:5060 "Noisy feedback tells: pid=6093 req_src_ip=50.16.169.79 req_src_port=55386 in_uri=sip:8355 at ideasip.com out_uri=sip:9797918005558355 at 208.97.25.12:5090 via_cnt==1"
Received INVITE response: SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.6.161.115:55386;received=50.16.169.79
Record-Route: <sip:208.97.25.11;ftag=433402263;lr=on>
From: 8355 <sip:8355 at 10.6.161.115>;tag=433402263
To: sip:8355 at ideasip.com;tag=as1dd4047f
Call-ID: 984943604 at 10.6.161.115
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:9797918005558355 at 208.97.25.12:5090>
Content-Type: application/sdp
Content-Length: 182
v=0
o=root 6571 6571 IN IP4 208.97.25.12
s=session
c=IN IP4 208.97.25.12
t=0 0
m=audio 19840 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
Opened URL "sip:8355 at ideasip.com", returning a SDP description:
v=0
o=- 984943604 0 IN IP4 10.6.161.115
s=playSIP session
c=IN IP4 10.6.161.115
t=0 0
m=audio 8000 RTP/AVP 0
Created receiver for "audio/PCMU" subsession (client ports 47616-47617)
Setup "audio/PCMU" subsession (client ports 47616-47617)
Created output file: "audio-PCMU-1"
Sending request: ACK sip:8355 at ideasip.com SIP/2.0
From: 8355 <sip:8355 at 10.6.161.115>;tag=433402263
Via: SIP/2.0/UDP 10.6.161.115:55386
To: sip:8355 at ideasip.com;tag=as1dd4047f
Call-ID: 984943604 at 10.6.161.115
CSeq: 1 ACK
Content-Length: 0
Started playing session
Receiving streamed data (signal with "kill -HUP 1502" or "kill -USR1 1502" to terminate)...
Got shutdown signal
begin_QOS_statistics
subsession audio/PCMU
num_packets_received 0
num_packets_lost 0
elapsed_measurement_time 8.000067
kBytes_received_total 0.000000
measurement_sampling_interval_ms 1000
kbits_per_second_min unavailable
kbits_per_second_ave unavailable
kbits_per_second_max unavailable
packet_loss_percentage_min 100.000000
packet_loss_percentage_ave 100.000000
packet_loss_percentage_max 100.000000
end_QOS_statistics
Sending request: BYE sip:8355 at ideasip.com SIP/2.0
From: 8355 <sip:8355 at 10.6.161.115>;tag=433402263
Via: SIP/2.0/UDP 10.6.161.115:55386
To: sip:8355 at ideasip.com;tag=as1dd4047f
Call-ID: 984943604 at 10.6.161.115
CSeq: 2 BYE
Content-Length: 0
fnadeau at ip-10-xxx-xxx-xxx:~/live/testProgs> ls -l audio-PCMU-1
-rw-r--r-- 1 fnadeau users 0 Oct 12 10:52 audio-PCMU-1
################################################################################################
Notice that the audio as 0 byte. The client port (from line Created receiver for "audio/PCMU" subsession (client ports 47616-47617)) is wrong. The SDP we send says we expect audio on port 8000.
We use an old build (don't ask) and get this as a result (we use the fix above to avoid segfault) :
################################################################################################
fnadeau at ip-10-xxx-xxx-xxx:~/Live555/testProgs> ./playSIP -Q -u nouser nopass ideasip.com sip:8355 at ideasip.com
Sending request: INVITE sip:8355 at ideasip.com SIP/2.0
From: nouser <sip:nouser at 10.6.161.115>;tag=146126391
Via: SIP/2.0/UDP 10.6.161.115:44716
To: sip:8355 at ideasip.com
Contact: sip:nouser at 10.6.161.115:44716
Call-ID: 1687940000 at 10.6.161.115
CSeq: 1 INVITE
Content-Type: application/sdp
User-Agent: playSIP (LIVE555 Streaming Media v2010.09.10)
Content-length: 116
v=0
o=- 1687940000 0 IN IP4 10.6.161.115
s=playSIP session
c=IN IP4 10.6.161.115
t=0 0
m=audio 8000 RTP/AVP 0
Received INVITE response: SIP/2.0 100 trying -- your call is important to us
From: nouser <sip:nouser at 10.6.161.115>;tag=146126391
Via: SIP/2.0/UDP 10.6.161.115:44716;received=50.16.169.79
To: sip:8355 at ideasip.com
Call-ID: 1687940000 at 10.6.161.115
CSeq: 1 INVITE
Server: Sip EXpress router (0.9.6 (i386/linux))
Content-Length: 0
Warning: 392 208.97.25.11:5060 "Noisy feedback tells: pid=6090 req_src_ip=50.16.169.79 req_src_port=44716 in_uri=sip:8355 at ideasip.com out_uri=sip:9797918005558355 at 208.97.25.12:5090 via_cnt==1"
Received INVITE response: SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.6.161.115:44716;received=50.16.169.79
Record-Route: <sip:208.97.25.11;ftag=146126391;lr=on>
From: nouser <sip:nouser at 10.6.161.115>;tag=146126391
To: sip:8355 at ideasip.com;tag=as5f442dd5
Call-ID: 1687940000 at 10.6.161.115
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:9797918005558355 at 208.97.25.12:5090>
Content-Type: application/sdp
Content-Length: 182
v=0
o=root 6571 6571 IN IP4 208.97.25.12
s=session
c=IN IP4 208.97.25.12
t=0 0
m=audio 19348 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
Opened URL "sip:8355 at ideasip.com", returning a SDP description:
v=0
o=- 1687940000 0 IN IP4 10.6.161.115
s=playSIP session
c=IN IP4 10.6.161.115
t=0 0
m=audio 8000 RTP/AVP 0
Created receiver for "audio/PCMU" subsession (client ports 8000-8001)
Scanning: [v=0]
Scanning: [o=root 6571 6571 IN IP4 208.97.25.12]
Scanning: [s=session]
Scanning: [c=IN IP4 208.97.25.12]
Scanning: [t=0 0]
Scanning: [m=audio 19348 RTP/AVP 0]
Scanning: [a=rtpmap:0 PCMU/8000]
Scanning: [a=silenceSupp:off - - - -]
Scanning: [a=ptime:20]
Scanning: [a=sendrecv]
Scanning: []
Setup "audio/PCMU" subsession (client ports 8000-8001)
Created output file: "audio-PCMU-1"
Sending request: ACK sip:8355 at ideasip.com SIP/2.0
From: nouser <sip:nouser at 10.6.161.115>;tag=146126391
Via: SIP/2.0/UDP 10.6.161.115:44716
To: sip:8355 at ideasip.com;tag=as5f442dd5
Call-ID: 1687940000 at 10.6.161.115
CSeq: 1 ACK
Content-length: 0
Started playing session
Receiving streamed data (signal with "kill -HUP 1504" or "kill -USR1 1504" to terminate)...
Got shutdown signal
begin_QOS_statistics
subsession audio/PCMU
num_packets_received 184
num_packets_lost 0
elapsed_measurement_time 10.001158
kBytes_received_total 29.440000
measurement_sampling_interval_ms 1000
kbits_per_second_min 0.000000
kbits_per_second_ave 23.549273
kbits_per_second_max 64.000064
packet_loss_percentage_min 0.000000
packet_loss_percentage_ave 0.000000
packet_loss_percentage_max 0.000000
inter_packet_gap_ms_min 16.417000
inter_packet_gap_ms_ave 19.888000
inter_packet_gap_ms_max 23.798000
end_QOS_statistics
Sending request: BYE sip:8355 at ideasip.com SIP/2.0
From: nouser <sip:nouser at 10.6.161.115>;tag=146126391
Via: SIP/2.0/UDP 10.6.161.115:44716
To: sip:8355 at ideasip.com;tag=as5f442dd5
Call-ID: 1687940000 at 10.6.161.115
CSeq: 2 BYE
Content-length: 0
fnadeau at ip-10-xxx-xxx-xxx:~/Live555/testProgs> ls -l audio-PCMU-1
-rw-r--r-- 1 fnadeau users 29440 Oct 12 10:53 audio-PCMU-1
################################################################################################
It is a slightly modified version, mostly on printout info.
This is run on EC2 as well (SUSE, but does not really matter). The SIP is publicaly available. The audio says : sorry you don't have enough credit to call this number. Could be better, but it is a test after all.
Since we are still using a old version, I couldn't port our modification to the latest version and test. I didn't have time to trace down the issue any further.
Frédéric Nadeau
De : live-devel-bounces at ns.live555.com [mailto:live-devel-bounces at ns.live555.com] De la part de Ross Finlayson
Envoyé : 11 octobre 2012 14:05
À : LIVE555 Streaming Media - development & use
Objet : Re: [Live-devel] PlaySIP SegFault
I've tried installing it onto an ubunutu 12.04 server on EC2 using the livemedia-utils package and also building from source on my iMac running 10.7, however both times I try to start it with a command "playsip sip:1234 at sip.sammachin.com<mailto:sip%3A1234 at sip.sammachin.com>" it just segfaults.
Any help welcome?
Thanks for the report. I've now installed a new version (2012.10.11) of the "LIVE555 Streaming Media" software that fixes this.
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
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