Asterisk Chan_sip Removing Ptime From DSP?

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Hi List

Just ran into another weird issue…

In Swiss Telephone Interconnection, ptime=20 is a requirement.

So on our SBC we enforce the presence of ptime=20 to avoid issues.

I have an asterisk with chan_sip in the LAB which behaves weirdly…

Inbound SDP audio part:

m=audio 15542 RTP/AVP 9 8 3 101
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:4hJMPvZmRA03KxLg8Hp+aHqeIhnmYBSQtwlT+Vkr a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

Outbound SDP audio part

m=audio 18802 RTP/AVP 8 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

Why is ptime missing on the outbound leg?

Our SBC answers 415

—
Mit freundlichen Grüssen

-Benoît Panizzon- @ HomeOffice und normal erreichbar
—
I m p r o W a r e A G – Leiter Commerce Kunden

One thought on - Asterisk Chan_sip Removing Ptime From DSP?

  • Also figured that out.

    For chan sip, specifying the ptime in the allow statement

    allow=g722:20,alaw:20 made the ptime header be present and the SBC
    happy.

    —
    Mit freundlichen Grüssen

    -Benoît Panizzon- @ HomeOffice und normal erreichbar
    —
    I m p r o W a r e A G – Leiter Commerce Kunden