CUBE - Skype for Business SIP trunk - Malformed SIP header

[juraj.konecny]

Hi everyone,

Im trying to configure SIP Trunk between S4B and CUBE. CUBE is next connected to PSTN provider.

When i try to call from S4B to mobile phone number  everything is ok. I can make a call. Problem is when i try to call back from mobile phone number to PSTN number. Profiles, translation rules and dial-peers are configured right in CUBE. Problem is in SIP header. Maybe i need to configure some voice class sip-profile. When i use WireShark to capture the packets i see this ICMP response for SIP INVITE packet:

ICMP  590  Destination unreachable (Port unreachable)

[Expert Info (Warn/Malformed): Header has no colon after the name]
[Header has no colon after the name]
[Severity level: Warn]
[Group: Malformed]

Header not terminated by empty line (CRLF)
[Expert Info (Warn/Malformed): Header not terminated by empty line (CRLF)]
[Header not terminated by empty line (CRLF)]
[Severity level: Warn]
[Group: Malformed]

Does anyone know, how to change the packet header?

[sanjaydevan]

You can create a sip profile and apply to a dial-peer going out to S4B

But before that please post the current header that you see

Regards

Devan

[juraj.konecny]

Hi Devon,

i have already solved the problem. The problem was in dial-peer when i used session transport udp. When i changed udp to tcp I partly solved the problem but i did not hear ringing tone in my mobile device.

This next command solved problem completely. In voice service voip sip i used rel1xx supported PRACK. 

This is ma dial-peer to S4B

dial-peer voice 7 voip
description to-Skype
translation-profile outgoing to_skype
destination-pattern  *called number*
session protocol sipv2
session target dns:*xxxxxxx.xx:5068*
session transport tcp
voice-class codec 10
voice-class sip bind control source-interface GigabitEthernet0/0
voice-class sip bind media source-interface GigabitEthernet0/0
dtmf-relay rtp-nte
no vad

I hope it helps someone.

Regards Juraj.