Debugging RTP with sipp and Wireshark
Mike Bradeen
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Did you know that sipp can play back RTP streams? A lot of long time users may not! Did you know that you can use
Did you know that sipp can play back RTP streams? A lot of long time users may not! Did you know that you can use
If you didn’t already know, both SIP and HTTP share the same digest authentication mechanism described all the way back in RFC-2069 “An Extension to HTTP
As mentioned in this post, Asterisk now supports the use of RFC4733 digits in common bitrates beyond 8kHz. At the end of the post, we
The ability to send SIP NOTIFY messages to endpoints and arbitrary uri’s via AMI and CLI has existed for some time within Asterisk. Changes from
Up until recently Asterisk only supported RFC 4733 RTP events when using 8KHz codecs like G.711. However, with this recent change, Asterisk now supports the
A recurring theme I’m seeing lately is people deploying VoIP, running into issues, and not approaching their issues from the perspective of taking all components
Overview If you’re familiar with Asterisk, you probably know that it uses a third-party project called pjproject. This is a major part of the PJSIP
Note: This new implementation is available as of Asterisk 18.22.0, 20.7.0, and 21.2.0. It’s been almost 4 years since STIR/SHAKEN support was first added to
I help out a lot across various avenues: mailing lists, forums, issue tracker, IRC, and more. I’ve noticed a trend recently where people aren’t verifying
chan_sip will no longer be included with Asterisk as of the release of version 21. Deprecated in version 17, chan_sip has been scheduled for removal for
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