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
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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
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
In a previous blog post we talked about using Asterisk’s uni-cast functionality as a bridge between the PSTN and an external service. This post entails
I should just be able to tell it what I want it to do. As the cost comes down for live transcription and the quality
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
If you keep an eye on the Asterisk gitlog, you may have seen some additions to app_voicemail. These changes include the ability to ‘show’ a
As you may have seen from a recent submission, it was recently found that there was a consistent delay when adding a channel into the
As part of the transition to Python3 at the end of last year, we introduced a Python Virtual Environment, or venv for the Asterisk TestSuite.
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