[Live-devel] playSIP crashed after getting 200 OK for invite

Vilaysak Thiphavong vthiphavong at arcatech.com
Mon Feb 8 05:29:45 PST 2016


Hi all,

I was trying to make call to my IP phone using playSIP with the following arguments

./playSIP -a -A8 sip:Dialogphone at 192.168.0.195

For any reason after answering a phone then playSIP crashed, Any idea please?

Below is the log:
root at systemcontroller testProgs]# ./playSIP -a -A 8 sip:Dialog at 192.168.0.195
Sending request: INVITE sip:Dialog at 192.168.0.195 SIP/2.0
From: Dialog <sip:Dialog at 192.168.0.147>;tag=541495922
Via: SIP/2.0/UDP 192.168.0.147:55086
Max-Forwards: 70
To: sip:Dialog at 192.168.0.195
Contact: sip:Dialog at 192.168.0.147:55086
Call-ID: 934922419 at 192.168.0.147
CSeq: 1 INVITE
Content-Type: application/sdp
User-Agent: playSIP (LIVE555 Streaming Media v2016.01.29)
Content-Length: 117

v=0
o=- 934922419 0 IN IP4 192.168.0.147
s=playSIP session
c=IN IP4 192.168.0.147
t=0 0
m=audio 8000 RTP/AVP 8

Received INVITE response: SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.147:55086
Call-ID: 934922419 at 192.168.0.147
Contact: "Dialog" <sip:Phone at 192.168.0.195>
CSeq: 1 INVITE
From: Dialog <sip:Dialog at 192.168.0.147>;tag=541495922
Supported: timer
To: sip:Dialog at 192.168.0.195;tag=00000f88-f0f0ff78
Server: ipDialog SipTone 1.2.0 rc Z_11 UAS
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,SUBSCRIBE,INFO,NOTIFY,REFER,MESSAGE
Content-Length: 0


Received INVITE response: SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.147:55086
Call-ID: 934922419 at 192.168.0.147
Contact: "Dialog" <sip:Phone at 192.168.0.195>
CSeq: 1 INVITE
From: Dialog <sip:Dialog at 192.168.0.147>;tag=541495922
Supported: timer
To: sip:Dialog at 192.168.0.195;tag=00000f88-f0f0ff78
Server: ipDialog SipTone 1.2.0 rc Z_11 UAS
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,SUBSCRIBE,INFO,NOTIFY,REFER,MESSAGE
Content-Type: application/sdp
Content-Length: 158

v=0
o=Phone 1010280478 1010280478 IN IP4 192.168.0.195
s=Sip Call
c=IN IP4 192.168.0.195
t=0 0
m=audio 5020 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20

Opened URL "sip:Dialog at 192.168.0.195", returning a SDP description:
v=0
o=- 934922419 0 IN IP4 192.168.0.147
s=playSIP session
c=IN IP4 192.168.0.147
t=0 0
m=audio 8000 RTP/AVP 8

Created receiver for "audio/PCMA" subsession (client ports 8000-8001)
Setup "audio/PCMA" subsession (client ports 8000-8001)
Segmentation fault (core dumped):1

[root at systemcontroller testProgs]#

Thanks,
Sak



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