RTP relay at Brekeke PBX
Difference between RTP relay on and off at Brekeke PBX
| RTP relay | ON | OFF |
| CPU usage |
High | Low |
| Traffic through BPBX server |
Large | Small |
| SIP devices / services used with BPBX | –Compatible with most SIP devices/services.
— Sending re-INVITE to User Agents (devices/services) does not occur unless SIP session timer is used. — For most SIP services in the market, trouble will be avoided by setting RTP relay ON at ARS rules |
–May not be compatible with certain SIP devices/services
–SIP devices/services which can handle changes in media information using re-INVITE. –Higher possibility of encountering issues with different SIP phones since RTP sessions are directly exchanged peer to peer. — To confirm interoperability, testing for all combination of devices is recommended |
| Keypad Commands | In-band, RFC2833, SIP-INFO (DTMF-relay) are available | Only SIP-INFO (DTMF-relay) is available |
| Codec Conversion |
Supported codecs conversion is handled by Brekeke PBX. User Agents can use different codecs. | Codecs conversion will not support. User Agents need to be connected using the same codec. |
About RTP Relay
Depending on your operating environment and its requirements, and to ensure proper SIP communications, the RTP relay through Brekeke PBX may be necessary.
For example:
- If you are using a SIP UA that does not support the SIP-INFO (DTMF-relay) method, and you wish to use keypad commands to activate Brekeke PBX features, such as Call Forwarding, Call Park, Call Recording, etc., RTP packets need to be relayed through the Media Server.
- If you are using SIP devices (SIP UA, SIP proxy server, etc.) that do not support changing RTP sender information by receiving re-INVITE requests, RTP packets need to be relayed through the Media Server.
- If you are using SIP devices that do not allow changing a voice codec during a call, select “on (G.711u only)” at the RTP relay setting or “0 (G.711u only)” at [Codec Priority] in the Brekeke PBX setting. Selecting “off” at the RTP relay setting may not work well depending on the SIP device. There may also be compatibility issues between different types of SIP devices. Thorough testing is recommended before deciding to set the RTP relay setting (which is not G.711u only) to “off” or to use a codec other than G.711u.
RTP relay settings can be set under several locations in Brekeke PBX: Options, ARS Settings and User Phone type Settings. The settings under User > [Phones] -> [Type] have the highest priority. When the default setting is defined in [Options] -> [Phone Type] and the Phone type is set in User -> [Phones] ->[Type], the settings under [ARS Settings] are applied. When User Phone type Settings and ARS Settings are set to their defaults, the settings at [Options] are applied.
RTP relay will be turned off only when both endpoints are set to RTP relay “off.” When one of the endpoints requires RTP relay, RTP will be relayed through Brekeke PBX. By increasing the amount of RTP packets handled by Brekeke PBX, the Brekeke PBX server load will increase. Even when the RTP relay setting is set to “off,” Brekeke PBX handles RTP for some features, such as Music On Hold and Voicemail. Thus, depending on the type of usage and environment, the maximum number of concurrent sessions may vary for each setting.
Brekeke PBX supports not only codec G.711 u-law, but also G.729 (Optional), G.711 A-law and iLBC. When the RTP relay setting is set to “on,” Brekeke PBX will do conversions between different codecs, so parties who are using devices with different codecs can talk each other without any problems.