PJSIP Status

Is the below still correct for PJSIP?

  • Created by [Tony Lewis], last modified on [08 Feb , 2016]

PJSIP is the emerging SIP technology in Asterisk. It provides additional functionality and features not present in the legacy chan_sip and over time it will become the predominant SIP technology. However, because of it’s youth and more extensive feature set, customers are likely to encounter more bugs and issues. We encourage customer who either need these features or want to help progress and mature the technology to run with PJSIP, even in production, with the acknowledgement that there will be more issues. The Asterisk and FreePBX (Sangoma) Development teams are fully behind PJSIP and will try to address all bugs and issues that arise from it. However, chan_sip still remains the mature SIP channel that should be used where stability is the most critical factor and tolerance for early adoption of new technologies can’t be tolerated.

No, it isn’t as far as I am concerned. The PJSIP stack can handle more scenarios and is much easier to configure. Sometimes it is the only option, like when a DNS hierarchy needs to get evaluated using NAPTR and SRV lookups.



id like to see us default to pjsip as the single channel driver soon :slight_smile:
keep the functionality to use both but hide it in an advanced setting ?

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Just as a data point - the Switchvox product has exclusively used chan_pjsip for… 3 years? 4 years? now. It doesn’t even build chan_sip or make it available. We ate our own dog food to improve it, fill in gaps, and test it with different other implementations.


IMO, it was not true in 2016 either.

but, meh. certainly not true now.

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