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
Overview When recording audio, it can be useful to split streams from one another. For example, if you are in a call with someone, you
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
Overview A new feature called Tenant ID (or multi-tenant identifier) was recently added to Asterisk. It doesn’t do anything on its own, but can be
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
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
Most SIP endpoints will qualify each other by sending OPTIONS back and forth to make sure they are there and can respond. Asterisk uses this mechanism
Asterisk 18.17.0 and 20.2.0 were released recently with support for PJSIP 2.13. This version of PJSIP includes an important change to deal with race conditions
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