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PortSIP VoIP SDK Manual for iOS
16.2
PortSIP features our newest, supported, quality-assured VoIP SDK used by Several hundred companies around the world for easy VoIP application develop and quality-assured code.
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Functions | |
(NSString *) | - PortSIPSDK::getVersion |
Get the current version number of the SDK. More... | |
(int) | - PortSIPSDK::enableRport: |
Enable/disable rport(RFC3581). More... | |
(int) | - PortSIPSDK::enableEarlyMedia: |
Enable/disable Early Media. More... | |
(int) | - PortSIPSDK::enableReliableProvisional: |
Enable/disable PRACK. More... | |
(int) | - PortSIPSDK::enable3GppTags: |
Enable/disable the 3Gpp tags, including "ims.icsi.mmtel" and "g.3gpp.smsip". More... | |
(void) | - PortSIPSDK::enableCallbackSendingSignaling: |
Enable/disable to callback the sent SIP messages. More... | |
(int) | - PortSIPSDK::setSrtpPolicy: |
Set the SRTP policy. More... | |
(int) | - PortSIPSDK::setRtpPortRange:maximumRtpAudioPort:minimumRtpVideoPort:maximumRtpVideoPort: |
Set the RTP ports range for audio and video streaming. More... | |
(int) | - PortSIPSDK::setRtcpPortRange:maximumRtcpAudioPort:minimumRtcpVideoPort:maximumRtcpVideoPort: |
Set the RTCP ports range for audio and video streaming. More... | |
(int) | - PortSIPSDK::enableCallForward:forwardTo: |
Enable call forwarding. More... | |
(int) | - PortSIPSDK::disableCallForward |
Disable the call forwarding. The SDK is not forwarding any incoming calls once this function is called. More... | |
(int) | - PortSIPSDK::enableSessionTimer:refreshMode: |
Allows to periodically refresh Session Initiation Protocol (SIP) sessions by sending INVITE requests repeatedly. More... | |
(int) | - PortSIPSDK::disableSessionTimer |
Disable the session timer. More... | |
(void) | - PortSIPSDK::setDoNotDisturb: |
Enable the "Do not disturb" to enable/disable. More... | |
(void) | - PortSIPSDK::enableAutoCheckMwi: |
Enable the CheckMwi to enable/disable. More... | |
(int) | - PortSIPSDK::setRtpKeepAlive:keepAlivePayloadType:deltaTransmitTimeMS: |
Enable or disable to send RTP keep-alive packet when the call is established. More... | |
(int) | - PortSIPSDK::setKeepAliveTime: |
Enable or disable to send SIP keep-alive packet. More... | |
(int) | - PortSIPSDK::setAudioSamples:maxPtime: |
Set the audio capturing sample. More... | |
(int) | - PortSIPSDK::addSupportedMimeType:mimeType:subMimeType: |
Set the SDK to receive the SIP message that includes special mime type. More... | |
- (NSString*) getVersion |
Get the current version number of the SDK.
- (int) enableRport: | (BOOL) | enable |
Enable/disable rport(RFC3581).
enable | Set to true to enable the SDK to support rport. By default it is enabled. |
- (int) enableEarlyMedia: | (BOOL) | enable |
Enable/disable Early Media.
enable | Set to true to enable the SDK to support Early Media. By default the Early Media is disabled. |
- (int) enableReliableProvisional: | (BOOL) | enable |
Enable/disable PRACK.
enable | Set to true to enable the SDK to support PRACK. By default the PRACK is disabled. |
- (int) enable3GppTags: | (BOOL) | enable |
Enable/disable the 3Gpp tags, including "ims.icsi.mmtel" and "g.3gpp.smsip".
enable | Set to true to enable the SDK to support 3Gpp tags. |
- (void) enableCallbackSendingSignaling: | (BOOL) | enable |
Enable/disable to callback the sent SIP messages.
enable | Set as true to enable to callback the sent SIP messages, or false to disable. Once enabled, the "onSendingSignaling" event will be triggered when the SDK sends a SIP message. |
- (int) setSrtpPolicy: | (SRTP_POLICY) | srtpPolicy |
Set the SRTP policy.
srtpPolicy | The SRTP policy. |
- (int) setRtpPortRange: | (int) | minimumRtpAudioPort | |
maximumRtpAudioPort: | (int) | maximumRtpAudioPort | |
minimumRtpVideoPort: | (int) | minimumRtpVideoPort | |
maximumRtpVideoPort: | (int) | maximumRtpVideoPort | |
Set the RTP ports range for audio and video streaming.
minimumRtpAudioPort | The minimum RTP port for audio stream. |
maximumRtpAudioPort | The maximum RTP port for audio stream. |
minimumRtpVideoPort | The minimum RTP port for video stream. |
maximumRtpVideoPort | The maximum RTP port for video stream. |
- (int) setRtcpPortRange: | (int) | minimumRtcpAudioPort | |
maximumRtcpAudioPort: | (int) | maximumRtcpAudioPort | |
minimumRtcpVideoPort: | (int) | minimumRtcpVideoPort | |
maximumRtcpVideoPort: | (int) | maximumRtcpVideoPort | |
Set the RTCP ports range for audio and video streaming.
minimumRtcpAudioPort | The minimum RTCP port for audio stream. |
maximumRtcpAudioPort | The maximum RTCP port for audio stream. |
minimumRtcpVideoPort | The minimum RTCP port for video stream. |
maximumRtcpVideoPort | The maximum RTCP port for video stream. |
- (int) enableCallForward: | (BOOL) | forBusyOnly | |
forwardTo: | (NSString *) | forwardTo | |
Enable call forwarding.
forBusyOnly | If this parameter is set as true, the SDK will forward all incoming calls when currently it's busy. If it's set as false, the SDK forward all incoming calls anyway. |
forwardTo | The target of call forwarding. It must in the format of sip:xxxx@. sip. ports ip.c om |
- (int) disableCallForward |
Disable the call forwarding. The SDK is not forwarding any incoming calls once this function is called.
- (int) enableSessionTimer: | (int) | timerSeconds | |
refreshMode: | (SESSION_REFRESH_MODE) | refreshMode | |
Allows to periodically refresh Session Initiation Protocol (SIP) sessions by sending INVITE requests repeatedly.
timerSeconds | The value of the refreshment interval in seconds. Minimum of 90 seconds required. |
refreshMode | Allow to set the session refreshment by UAC or UAS: SESSION_REFERESH_UAC or SESSION_REFERESH_UAS; |
- (int) disableSessionTimer |
Disable the session timer.
- (void) setDoNotDisturb: | (BOOL) | state |
Enable the "Do not disturb" to enable/disable.
state | If it is set to true, the SDK will reject all incoming calls anyway. |
- (void) enableAutoCheckMwi: | (BOOL) | state |
Enable the CheckMwi to enable/disable.
state | If it is set to true, the SDK will check Mwi automatically. |
- (int) setRtpKeepAlive: | (BOOL) | state | |
keepAlivePayloadType: | (int) | keepAlivePayloadType | |
deltaTransmitTimeMS: | (int) | deltaTransmitTimeMS | |
Enable or disable to send RTP keep-alive packet when the call is established.
state | Set as true to allow to send the keep-alive packet during the conversation. |
keepAlivePayloadType | The payload type of the keep-alive RTP packet. It's usually set to 126. |
deltaTransmitTimeMS | The keep-alive RTP packet sending interval, in milliseconds. Recommended value ranges 15000 - 300000. |
- (int) setKeepAliveTime: | (int) | keepAliveTime |
Enable or disable to send SIP keep-alive packet.
keepAliveTime | This is the SIP keep-alive time interval in seconds. By setting to 0, the SIP keep-alive will be disabled. Recommended value is 30 or 50. |
- (int) setAudioSamples: | (int) | ptime | |
maxPtime: | (int) | maxPtime | |
Set the audio capturing sample.
ptime | It should be a multiple of 10 between 10 - 60 (with 10 and 60 inclusive). |
maxPtime | For the "maxptime" attribute, it should be a multiple of 10 between 10 - 60 (with 10 and 60 inclusive). It cannot be less than "ptime". |
- (int) addSupportedMimeType: | (NSString *) | methodName | |
mimeType: | (NSString *) | mimeType | |
subMimeType: | (NSString *) | subMimeType | |
Set the SDK to receive the SIP message that includes special mime type.
methodName | Method name of the SIP message, such as INVITE, OPTION, INFO, MESSAGE, UPDATE, ACK etc. For more details please refer to the RFC3261. |
mimeType | The mime type of SIP message. |
subMimeType | The sub mime type of SIP message. |