R0106-HP MSR Router Series Voice Configuration Guide(V7)

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Configuring SRTP for SIP calls
The differences between the srtp and srtp fallback commands are as follows:
• With the srtp command configured:
{ The device includes crypto and RTP/SAVP parameters in outgoing INVITE requests and
disconnects the call after receiving a 488 response.
{ The device can only accept calls using SRTP.
• With the srtp fallback command configured:
{ The device includes crypto and RTP/SAVP parameters in outgoing INVITE requests and
retransmits INVITE requests with RTP/AVP parameters after receiving a 488 response.
{ The device can accept calls using SRTP or RTP. SRTP is preferred for media stream protocol
negotiation. If the negotiation fails, RTP is used.
You can configure the srtp command globally (in SIP view) and for a specific VoIP entity (in VoIP entity
view). The configuration in VoIP entity view takes precedence over the global configuration. A VoIP entity
uses the global configuration only when the srtp command is not configured in VoIP entity view.
Enabling SRTP globally
Ste
p
Command Remarks
1. Enter system view.
system-view N/A
2. Enter voice view.
voice-setup N/A
3. Enter SIP view.
sip N/A
4. Enable SRTP globally.
srtp [ fallback ] By default, RTP is enabled globally.
Enabling SRTP on a VoIP entity
Ste
p
Command Remarks
1. Enter system view.
system-view N/A
2. Enter voice view.
voice-setup N/A
3. Enter SIP view.
sip N/A
4. Enter VoIP entity view.
entity entity-number voip N/A
5. Enable SRTP for the VoIP entity.
srtp [ fallback ] By default, RTP is used.
Specifying a URL scheme
The device provides two URL schemes: SIP and SIP secure (SIPS). To ensure end-to-end transmission
security, specify the SIPS scheme. This scheme requires using TLS as the transport protocol.
You can configure the URL scheme both globally (by using the url command in SIP view) and for a
specific VoIP entity (by using the voice-class sip url command in VoIP entity view). The configuration in