U 8.8.3.80:62003 -> 8.8.3.48:5060 INVITE sip:[email protected] SIP/2.0. Via: SIP/2.0/UDP 192.168.0.111:5060;branch=z9hG4bK31390;rport. From: 1001 <sip:[email protected]>;tag=2824524117. To: 1000 <sip:[email protected]>. Call-ID: [email protected]. CSeq: 1 INVITE. Contact: <sip:[email protected]:5060>. max-forwards: 70. supported: 100rel. user-agent: Voip Phone 1.0. Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, SUBSCRIBE, PRACK, UPDATE. Content-Type: application/sdp. Content-Length: 295 . . v=0. o=sdp_admin 30472538 21739392 IN IP4 192.168.0.111. s=A conversation. c=IN IP4 192.168.0.111. t=0 0. m=audio 10050 RTP/AVP 0 4 18 8 101. a=rtpmap:0 PCMU/8000. a=rtpmap:4 G723/8000. a=rtpmap:18 G729/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 8.8.3.48:5060 -> 8.8.3.80:62003 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 192.168.0.111:5060;branch=z9hG4bK31390;rport...
Building Telephony Systems with OpenSER
Building Telephony Systems with OpenSER
Overview of this book
Table of Contents (18 chapters)
Building Telephony Systems with OpenSER
Credits
About the Author
About the Reviewers
Preface
Free Chapter
Introduction to SIP
The SIP Express Router
OpenSER Installation
OpenSER Standard Configuration
Adding Authentication with MySQL
Building the User Portal with SerMyAdmin
Connectivity to the PSTN
Call Forward and Voice Mail
SIP NAT Traversal
OpenSER Accounting and Billing
Troubleshooting Tools
After Words
Index
Customer Reviews