Ptime

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Asterisk Users 2 Comments

We have a customer with a system rejecting calls from Asterisk. It’s indicating the ptime is 60, but the system admin is saying they only support 20.

They are running asterisk 16.2.1 and using chan_sip Is there a way to specify this with chan_sip?

Also, for my own curiosity, is there a way to specify this with PJSIP? (Trying to migrate customers to PJSIP, but we are holding until asterisk 16.6.0 for a PJSIP REFER change that has been merged).

Have a great day!
Dan

2 thoughts on - Ptime

  • The ptime is specified the same way in both chan_sip and chan_pjsip, with the codec. For example:

    allow=ulaw:20

    I don’t know why it would have been offering 60ms though. What codecs were allowed?

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  • Thank you Joshua

    We are specifying allow=ulaw

    We ran a capture on the asterisk side and it is not specifying a ptime value. In the INVITE, asterisk sends a maxptime=150. Then, the Avaya switch rejects the call.

    We are asking for additional information from the customer and why they think we are sending ptime=60.

    I just ran my own tests with chan_sip and I don’t think ptime in the codec works (at least not for outbound). With or without the ptime value (allow=ulaw or allow=ulaw:20) it is not sending ptime in the INVITE packet. I also tried changing the ptime in the codec to 40 (just in case it doesn’t send if it matches the default) but it also didn’t send it. The ptime is specified the same way in both chan_sip and chan_pjsip, with the codec. For example:

    allow=ulaw:20

    I don’t know why it would have been offering 60ms though. What codecs were allowed?