Hello Richard,

Your solution was half the fix.

I set via-branch=next on the initial INVITE and I set via-branch=1 on the reply.

For anyone else reading this, via-branch=1 is needed on the reply so that the FIRST via header is used to match the via header from via-branch=next.

via-branch=auto+via-branch=auto => This is good if the branching happened on the previous SIP Proxy and the current proxy is handling both branches.

via-branch=next+via-branch=1 => This is good if the branching happens on the current SIP Proxy.

With this, the audio works perfectly and is no longer dropped.

Thank you!

David

On Thu, Aug 7, 2025 at 7:59 AM Richard Fuchs via sr-users <sr-users@lists.kamailio.org> wrote:

On 06/08/2025 20.47, David Tek via sr-users wrote:

When doing local forking, how can I set the correct via-branch value for each leg in rtpengine_manage()?

Have you tried with `via-branch=next` ?

Cheers

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