# Issues with PJSIP and (Opus | g722)

**URL:** https://forums.vitalpbx.org/t/issues-with-pjsip-and-opus-g722/397
**Category:** VitalPBX
**Created:** [September 6, 2021, 2:11pm UTC](https://forums.vitalpbx.org/t/issues-with-pjsip-and-opus-g722/397 "2021-09-06T14:11:31Z")
**Posts on this page:** 4
**Page:** 1

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### Author: ![tsf\_it](https://forums.vitalpbx.org/letter_avatar_proxy/v4/letter/t/ce7236/32.png) [@tsf\_it](https://forums.vitalpbx.org/u/tsf_it)
#### Post date: [September 6, 2021, 2:11pm UTC](https://forums.vitalpbx.org/t/issues-with-pjsip-and-opus-g722/397/1 "2021-09-06T14:11:31Z")

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Dear all,

We are testing testing VitalPBX 3.1.1-1 . We managed to get everything we needed up and running using chan\_sip (local calls using Opus, call through our trunk using g.722), incomming and outgoing calls, queues etc…

We are equiped with Yealink T53 SIP phones and have a 10 Mbps symmetrical dedicated line for VOIP.

We have been working with pure asterisk and chan\_sip for years.

Our Yealink phones are using PJSIP as SIP stack for their firmwares, and also PJSIP is the stack privileged now by the Asterisk project so we gave a try to PJSIP. But this is where we face issues when using Opus.  
Something easy to reproduce is configuring a new extension and association a new device using a PJSIP profile (on which we have enabled G722 and Opus and set Opus as prefered codec, followed by G722). If we call the voicemail, we don’t hear. If I use a Linphone device (Opus enabled), I hear a very choppy announce. It’s a little bit better for G722 but not so much. However, if I set ulaw or alaw as prefered codec, everything works correctly.

This happens only when using the pjsip chan driver, the same tests with sip chan driver give normal results.

Clément

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### Author: ![miguel](https://forums.vitalpbx.org/user_avatar/forums.vitalpbx.org/miguel/32/415_2.png) [@miguel](https://forums.vitalpbx.org/u/miguel)
#### Post date: [September 7, 2021, 2:04am UTC](https://forums.vitalpbx.org/t/issues-with-pjsip-and-opus-g722/397/2 "2021-09-07T02:04:34Z")

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Have you asked in the Asterisk forum about this?

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### Author: ![tsf\_it](https://forums.vitalpbx.org/letter_avatar_proxy/v4/letter/t/ce7236/32.png) [@tsf\_it](https://forums.vitalpbx.org/u/tsf_it)
#### Post date: [September 9, 2021, 9:45am UTC](https://forums.vitalpbx.org/t/issues-with-pjsip-and-opus-g722/397/3 "2021-09-09T09:45:16Z")

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Actually, the problem came from the “Asymmetric RTP” option which was set to “Yes” in the default PJSIP profile.

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### Author: ![system](https://forums.vitalpbx.org/uploads/default/original/1X/672b458a1d454755270c2c16adc5316b641c3c6a.png) [@system](https://forums.vitalpbx.org/u/system)
#### Post date: [October 9, 2021, 9:45am UTC](https://forums.vitalpbx.org/t/issues-with-pjsip-and-opus-g722/397/4 "2021-10-09T09:45:43Z")

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