diff options
| author | Gerald Begumisa <ben_g@users.sourceforge.net> | 2007-09-21 17:26:05 +0000 |
|---|---|---|
| committer | Gerald Begumisa <ben_g@users.sourceforge.net> | 2007-09-21 17:26:05 +0000 |
| commit | 5a418de8f295d76e16873a0f1a8314e2a9fa6745 (patch) | |
| tree | 10cc23c3fecf6ff2d876c58503885f4b9e65a286 /orkaudio/audiocaptureplugins/voip/VoIp.cpp | |
| parent | 92876587193f588bb1fd34853861f3fedea25b6a (diff) | |
Added support to report RTP events in SIP sessions
git-svn-id: https://oreka.svn.sourceforge.net/svnroot/oreka/trunk@494 09dcff7a-b715-0410-9601-b79a96267cd0
Diffstat (limited to 'orkaudio/audiocaptureplugins/voip/VoIp.cpp')
| -rw-r--r-- | orkaudio/audiocaptureplugins/voip/VoIp.cpp | 31 |
1 files changed, 30 insertions, 1 deletions
diff --git a/orkaudio/audiocaptureplugins/voip/VoIp.cpp b/orkaudio/audiocaptureplugins/voip/VoIp.cpp index d85d05c..2d3faba 100644 --- a/orkaudio/audiocaptureplugins/voip/VoIp.cpp +++ b/orkaudio/audiocaptureplugins/voip/VoIp.cpp @@ -1031,7 +1031,7 @@ bool TryRtp(EthernetHeaderStruct* ethernetHeader, IpHeaderStruct* ipHeader, UdpH u_short dest = ntohs(udpHeader->dest); if(!(ntohs(udpHeader->source)%2) && !(ntohs(udpHeader->dest)%2) || DLLCONFIG.m_rtpDetectOnOddPorts) // udp ports usually even { - if((rtpHeader->pt <= 34 && rtpHeader->pt != 13) || rtpHeader->pt == 97 || rtpHeader->pt == 98) // pt=34 is H263 and is the last possible valid codec + if((rtpHeader->pt <= 34 && rtpHeader->pt != 13) || rtpHeader->pt == 97 || rtpHeader->pt == 98 || rtpHeader->pt > 98) // pt=34 is H263 and is the last possible valid codec, pt > 98 is for the case of SIP telephone-event { // pt=13 is CN (Comfort Noise) result = true; u_char* payload = (u_char *)rtpHeader + sizeof(RtpHeaderStruct); @@ -1470,6 +1470,7 @@ bool TrySipInvite(EthernetHeaderStruct* ethernetHeader, IpHeaderStruct* ipHeader char* audioField = NULL; char* connectionAddressField = NULL; char* attribSendonly = memFindAfter("a=sendonly", (char*)udpPayload, sipEnd); + char* rtpmapAttribute = memFindAfter("\na=rtpmap:", (char*)udpPayload, sipEnd); if(fromField) { @@ -1583,6 +1584,34 @@ bool TrySipInvite(EthernetHeaderStruct* ethernetHeader, IpHeaderStruct* ipHeader } } + if(rtpmapAttribute) + { + CStdString rtpPayloadType, nextToken; + char *nextStep = NULL; + + while(rtpmapAttribute && rtpmapAttribute < sipEnd) + { + GrabTokenSkipLeadingWhitespaces(rtpmapAttribute, sipEnd, rtpPayloadType); + nextToken.Format("%s ", rtpPayloadType); + nextStep = memFindAfter((char*)nextToken.c_str(), rtpmapAttribute, sipEnd); + + /* We need our "nextStep" to contain at least the length + * of the string "telephone-event", 15 characters */ + if(nextStep && ((sipEnd - nextStep) >= 15)) + { + if(strncasecmp(nextStep, "telephone-event", 15) == 0) + { + /* Our DTMF packets are indicated using + * the payload type rtpPayloadType */ + info->m_telephoneEventPayloadType = rtpPayloadType; + break; + } + } + + rtpmapAttribute = memFindAfter("\na=rtpmap:", rtpmapAttribute, sipEnd); + } + } + if((unsigned int)info->m_fromRtpIp.s_addr == 0) { // In case connection address could not be extracted, use SIP invite sender IP address |