Cisco SIP Router - getting the loopback to be the SIP peer

I was trying to bring up a SIP over IPSec peering on a CIsco ISR4451-X. This is to complement and already existing
SIP router (CUBE I guess they call it). Anyhow I had a call not connecting and I ran debug ccsip call. One weird thing
I noticed is that the Source IP Media address address on the external Gig 0/1 interface of the router.
The loopback is and that's what I'm expecting to see for Source IP Address (Media). What determines
what interface/address gets used for Source IP Address (Media)? Thank you.

001812: Dec 17 2018 23:26:18.025 PST: //747/E74BC54D82EC/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : g711ulaw
Negotiated Codec Bytes   : 160
Nego. Codec payload      : 0 (tx), 0 (rx)
Negotiated Dtmf-relay    : 6
Dtmf-relay Payload       : 101 (tx), 101 (rx)
Source IP Address (Media):        
Source IP Port    (Media): 17876
Destn  IP Address (Media):
Destn  IP Port    (Media): 55414
Orig Destn IP Address:Port (Media): [ - ]:0
amigan_99Network EngineerAsked:
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Timothy EstesSr. Voice EngineerCommented:
Letting the router bind sip control and media is never a good idea. You're right that you should bind it yourself. Binding "control" controls what the source IP is for the actual SIP traffic on TCP or UDP 5060 and 5061. "Media" determines what the source IP is for the RTP traffic.

Device(config)# voice service voip
Device(conf-voi-serv)# sip
Device(conf-serv-sip)# bind control source-interface FastEthernet0/0
Device(conf-serv-sip)# exit
Device(config)# voice service voip
Device(conf-voi-serv)# sip
Device(conf-serv-sip)# bind media source-address ipv4
Device(conf-serv-sip)# exit

You may have to bind both sides, (TSP and the SIP trunk to CUCM). This is done at the dial-peer level.

dial-peer voice 2 voip
description SIP TRUNK OUTGOING  
preference 1
destination-pattern 99T
session protocol sipv2
session target ipv4:172.X.X.X:5060
session transport udp
voice-class sip bind control source-interface Loopback1000
voice-class sip bind media source-interface Loopback1000
dtmf-relay rtp-nte
no dtmf-interworking
codec g711alaw
no vad

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amigan_99Network EngineerAuthor Commented:
Oh excellent. That's it. Thanks so much.
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