diff --git a/CHANGES.md b/CHANGES.md index 301ff8de2..b895aee8f 100644 --- a/CHANGES.md +++ b/CHANGES.md @@ -1,3 +1,17 @@ +Changes in 3.6.0 +================ + +* Removed unused RTPStream code concerning video streams. Also + consolidated the rtpstream audio port usage to reuse the global + `[media_port]` instead of the `[rtpstream_audio_port]`. + Also the `-min_rtp_port` and `-max_rtp_port` options have been + removed. Advantages: cleaner code, fewer scenario variables. + Drawbacks: possible ICMP port unreachable messages for RCTP and video. + Also, no easy way to discern different streams if you want to bombard + a single UAS with multiple RTP streams. (Issue #192, reported by + @atsakiridis.) + + Bugs fixed in 3.5.1 =================== diff --git a/include/sipp.hpp b/include/sipp.hpp index e53476617..c923a6d53 100644 --- a/include/sipp.hpp +++ b/include/sipp.hpp @@ -255,8 +255,6 @@ extern int tcp_readsize _DEFVAL(65535); extern int hasMedia _DEFVAL(0); #endif #ifdef RTP_STREAM -extern int min_rtp_port _DEFVAL(DEFAULT_MIN_RTP_PORT); -extern int max_rtp_port _DEFVAL(DEFAULT_MAX_RTP_PORT); extern int rtp_default_payload _DEFVAL(DEFAULT_RTP_PAYLOAD); extern int rtp_tasks_per_thread _DEFVAL(DEFAULT_RTP_THREADTASKS); extern int rtp_buffsize _DEFVAL(65535); diff --git a/regress/github-#0196/beep_1sec_50x160b.alaw b/regress/github-#0196/beep_1sec_50x160b.alaw new file mode 100644 index 000000000..29f5b258e Binary files /dev/null and b/regress/github-#0196/beep_1sec_50x160b.alaw differ diff --git a/regress/github-#0196/run b/regress/github-#0196/run new file mode 100755 index 000000000..bd2311f42 --- /dev/null +++ b/regress/github-#0196/run @@ -0,0 +1,32 @@ +#!/bin/sh +# This regression test is a part of SIPp. +# Author: Walter Doekes, OSSO B.V., 2016 +# +# Similar to github-#0192, it checks whether the rtpstream source port +# is correct, but this time uses the [media_port] instead of a special +# [rtpstream_audio_port]. +# +. "`dirname "$0"`/../functions"; init + +uac_media_port=5072 # client port +uas_media_port=5071 # server port + +udplisten $uas_media_port >udplisten.log & +udplisten_job=$! +trap "kill -9 $udplisten_job 2>/dev/null" EXIT + +sippbg -sf uas.xml -p 5070 -key custom_media_port $uas_media_port +sippfg -m 1 -sf uac.xml 127.0.0.1:5070 -mp $uac_media_port \ + -timeout 5 -timeout_error >/dev/null 2>&1 +status=$? + +test $status -ne 0 && fail "SIPp UAC job failed" + +port=`sed -e '/^Connection from/!d;s/.*://' udplisten.log` +if test "$port" = "$uac_media_port"; then + ok +elif test -n "$port"; then + fail "got RTP from wrong source port $port" +else + fail "got no RTP at all" +fi diff --git a/regress/github-#0196/uac.xml b/regress/github-#0196/uac.xml new file mode 100644 index 000000000..4640494b0 --- /dev/null +++ b/regress/github-#0196/uac.xml @@ -0,0 +1,80 @@ + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + diff --git a/regress/github-#0196/uas.xml b/regress/github-#0196/uas.xml new file mode 100644 index 000000000..758d4ea87 --- /dev/null +++ b/regress/github-#0196/uas.xml @@ -0,0 +1,78 @@ + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + diff --git a/src/call.cpp b/src/call.cpp index 6977e8b7c..2783610c0 100644 --- a/src/call.cpp +++ b/src/call.cpp @@ -2182,26 +2182,10 @@ char* call::createSendingMessage(SendingMessage *src, int P_index, char *msg_buf break; } #ifdef RTP_STREAM - case E_Message_RTPStream_Audio_Port: - { - int temp_audio_port= rtpstream_get_audioport (&rtpstream_callinfo); - if (!temp_audio_port) { - /* Make this a warning instead? */ - ERROR("cannot assign a free audio port to this call - using 0 for [rtpstream_audio_port]"); - } - dest += snprintf(dest, left, "%d",temp_audio_port); - } - break; - case E_Message_RTPStream_Video_Port: - { - int temp_video_port= rtpstream_get_videoport (&rtpstream_callinfo); - if (!temp_video_port) { - /* Make this a warning instead? */ - ERROR("cannot assign a free video port to this call - using 0 for [rtpstream_video_port]"); - } - dest += snprintf(dest, left, "%d",temp_video_port); - } - break; + case E_Message_RTPStream_Audio_Port: /* DEPRECATED */ + case E_Message_RTPStream_Video_Port: /* DEPRECATED */ + dest += sprintf(dest, "%u", media_port); + break; #endif case E_Message_Media_IP_Type: dest += snprintf(dest, left, "%s", (media_ip_is_ipv6 ? "6" : "4")); diff --git a/src/message.cpp b/src/message.cpp index 5e579545e..0d86aada1 100644 --- a/src/message.cpp +++ b/src/message.cpp @@ -62,8 +62,9 @@ struct KeywordMap SimpleKeywords[] = { {"auto_media_port", E_Message_Auto_Media_Port }, #endif #ifdef RTP_STREAM - {"rtpstream_audio_port", E_Message_RTPStream_Audio_Port }, - {"rtpstream_video_port", E_Message_RTPStream_Video_Port }, + /* Legacy since 3.6-dev. Actually uses media_port. */ + {"rtpstream_audio_port", E_Message_RTPStream_Audio_Port }, + {"rtpstream_video_port", E_Message_RTPStream_Video_Port }, #endif {"media_port", E_Message_Media_Port }, {"media_ip_type", E_Message_Media_IP_Type }, diff --git a/src/rtpstream.cpp b/src/rtpstream.cpp index e0d5d1af5..a0bcec480 100644 --- a/src/rtpstream.cpp +++ b/src/rtpstream.cpp @@ -39,14 +39,12 @@ static void debugprint(const char *Format, ...) #define MAX_UDP_RECV_BUFFER 8192 #define TI_NULL_AUDIOIP 0x01 -#define TI_NULL_VIDEOIP 0x02 -#define TI_NULLIP (TI_NULL_AUDIOIP|TI_NULL_VIDEOIP) +#define TI_NULLIP (TI_NULL_AUDIOIP) #define TI_PAUSERTP 0x04 #define TI_ECHORTP 0x08 /* Not currently implemented */ #define TI_KILLTASK 0x10 -#define TI_RECONNECTSOCKET 0x20 #define TI_PLAYFILE 0x40 -#define TI_CONFIGFLAGS (TI_KILLTASK|TI_RECONNECTSOCKET|TI_PLAYFILE) +#define TI_CONFIGFLAGS (TI_KILLTASK|TI_PLAYFILE) struct rtp_header_t { @@ -90,15 +88,9 @@ struct taskentry_t int new_timeticks_per_packet; /* sockets for audio/video rtp_rtcp */ int audio_rtp_socket; - int audio_rtcp_socket; - int video_rtp_socket; - int video_rtcp_socket; /* rtp peer address structures */ struct sockaddr_storage remote_audio_rtp_addr; - struct sockaddr_storage remote_audio_rtcp_addr; - struct sockaddr_storage remote_video_rtp_addr; - struct sockaddr_storage remote_video_rtcp_addr; /* we will have a mutex per call. should we consider refactoring to */ /* share mutexes across calls? makes the per-call code more complex */ @@ -143,94 +135,16 @@ unsigned int global_ssrc_id= 0xCA110000; /* code checked */ static void rtpstream_free_taskinfo(taskentry_t* taskinfo) { - if (taskinfo) { - /* close sockets associated with this call */ - if (taskinfo->audio_rtp_socket!=-1) { - close (taskinfo->audio_rtp_socket); - } - if (taskinfo->audio_rtcp_socket!=-1) { - close (taskinfo->audio_rtcp_socket); - } - if (taskinfo->video_rtp_socket!=-1) { - close (taskinfo->video_rtp_socket); + if (taskinfo) { + /* cleanup pthread library structure */ + pthread_mutex_destroy(&(taskinfo->mutex)); + free(taskinfo); } - if (taskinfo->video_rtcp_socket!=-1) { - close (taskinfo->video_rtcp_socket); - } - - /* cleanup pthread library structure */ - pthread_mutex_destroy(&(taskinfo->mutex)); - - free (taskinfo); - } } /* code checked */ static void rtpstream_process_task_flags(taskentry_t* taskinfo) { - if (taskinfo->flags&TI_RECONNECTSOCKET) { - int remote_addr_len; - int rc; - - remote_addr_len = media_ip_is_ipv6 ? sizeof(struct sockaddr_in6) : sizeof(struct sockaddr_in); - - /* enter critical section to lock address updates */ - /* may want to leave this out -- low chance of race condition */ - pthread_mutex_lock (&(taskinfo->mutex)); - - /* If we have valid ip and port numbers for audio rtp stream */ - if (!(taskinfo->flags & TI_NULL_AUDIOIP)) - { - if (taskinfo->audio_rtcp_socket != -1) { - rc = connect(taskinfo->audio_rtcp_socket, (struct sockaddr *)&taskinfo->remote_audio_rtcp_addr, - remote_addr_len); - if (rc < 0) { - debugprint("closing audio rtcp socket %d due to error %d in rtpstream_process_task_flags taskinfo=%p\n", - taskinfo->audio_rtcp_socket, errno, taskinfo); - close(taskinfo->audio_rtcp_socket); - taskinfo->audio_rtcp_socket = -1; - } - } - - if (taskinfo->audio_rtp_socket != -1) { - rc = connect(taskinfo->audio_rtp_socket, (struct sockaddr *)&taskinfo->remote_audio_rtp_addr, remote_addr_len); - if (rc < 0) { - debugprint("closing audio rtp socket %d due to error %d in rtpstream_process_task_flags taskinfo=%p\n", - taskinfo->audio_rtp_socket, errno, taskinfo); - close(taskinfo->audio_rtp_socket); - taskinfo->audio_rtp_socket = -1; - } - } - } - - /* If we have valid ip and port numbers for video rtp stream */ - if (!(taskinfo->flags & TI_NULL_VIDEOIP)) - { - if (taskinfo->video_rtcp_socket != -1) { - rc = connect(taskinfo->video_rtcp_socket, (struct sockaddr *)&taskinfo->remote_video_rtcp_addr, - remote_addr_len); - if (rc < 0) { - debugprint("closing video rtcp socket %d due to error %d in rtpstream_process_task_flags taskinfo=%p\n", - taskinfo->video_rtcp_socket, errno, taskinfo); - close(taskinfo->video_rtcp_socket); - taskinfo->video_rtcp_socket = -1; - } - } - if (taskinfo->video_rtp_socket != -1) { - rc = connect(taskinfo->video_rtp_socket, (struct sockaddr *)&taskinfo->remote_video_rtp_addr, - remote_addr_len); - if (rc < 0) { - debugprint("closing video rtp socket %d due to error %d in rtpstream_process_task_flags taskinfo=%p\n", - taskinfo->video_rtp_socket, errno, taskinfo); - close(taskinfo->video_rtp_socket); - taskinfo->video_rtp_socket = -1; - } - } - } - - taskinfo->flags&= ~TI_RECONNECTSOCKET; - pthread_mutex_unlock (&(taskinfo->mutex)); - } if (taskinfo->flags&TI_PLAYFILE) { /* copy playback information */ taskinfo->loop_count= taskinfo->new_loop_count; @@ -267,41 +181,11 @@ static unsigned long rtpstream_playrtptask(taskentry_t *taskinfo, unsigned long next_wake = timenow_ms + 100; /* default next wakeup time */ - if (taskinfo->audio_rtcp_socket != -1) { - /* just keep listening on rtcp socket (is this really required?) - ignore any errors */ - while ((rc = recv(taskinfo->audio_rtcp_socket, udp.buffer, sizeof(udp), 0)) >= 0) { - /* - * rtpstream_bytes_in+= rc; - */ - } - } - - if (taskinfo->video_rtp_socket != -1) { - /* just keep listening on rtp socket (is this really required?) - ignore any errors */ - while ((rc = recv(taskinfo->video_rtp_socket, udp.buffer, sizeof(udp), 0)) >= 0) { - /* - * rtpstream_bytes_in += rc; - */ - } - } - - if (taskinfo->video_rtcp_socket != -1) { - /* just keep listening on rtcp socket (is this really required?) - ignore any errors */ - while ((rc = recv(taskinfo->video_rtcp_socket, udp.buffer, sizeof(udp), 0)) >= 0) { - /* - * rtpstream_bytes_in+= rc; - */ - } - } - if (taskinfo->audio_rtp_socket != -1) { - /* this is temp code - will have to reorganize if/when we include echo functionality */ - /* just keep listening on rtcp socket (is this really required?) - ignore any errors */ - while ((rc = recv(taskinfo->audio_rtp_socket, udp.buffer, sizeof(udp), 0)) >= 0) { - /* for now we will just ignore any received data or receive errors */ - /* separate code path for RTP echo */ - rtpstream_bytes_in += rc; - } + /* if/when we include echo functionality, we'll have to read + * from the audio_rtp_socket too, and check by peer address if + * it is "our" traffic */ + /* are we playing back an audio file? */ if (taskinfo->loop_count) { target_timestamp = timenow_ms * taskinfo->timeticks_per_ms; @@ -311,6 +195,7 @@ static unsigned long rtpstream_playrtptask(taskentry_t *taskinfo, unsigned long taskinfo->last_timestamp = target_timestamp; } if (taskinfo->last_timestamp < target_timestamp) { + /* need to send rtp payload - build rtp packet header... */ udp.hdr.flags = htons(0x8000 | taskinfo->payload_type); udp.hdr.seq = htons(taskinfo->seq); @@ -327,8 +212,15 @@ static unsigned long rtpstream_playrtptask(taskentry_t *taskinfo, unsigned long memcpy(udp.buffer + sizeof(rtp_header_t) + taskinfo->file_bytes_left, taskinfo->file_bytes_start, taskinfo->bytes_per_packet-taskinfo->file_bytes_left); } + /* now send the actual packet */ - rc = send(taskinfo->audio_rtp_socket, udp.buffer, taskinfo->bytes_per_packet + sizeof(rtp_header_t), 0); + size_t packet_len = taskinfo->bytes_per_packet + sizeof(rtp_header_t); + socklen_t remote_addr_len = (media_ip_is_ipv6 ? + sizeof(struct sockaddr_in6) : + sizeof(struct sockaddr_in)); + rc = sendto(taskinfo->audio_rtp_socket, udp.buffer, packet_len, 0, + (struct sockaddr*)&taskinfo->remote_audio_rtp_addr, remote_addr_len); + if (rc < 0) { /* handle sending errors */ if (errno == EAGAIN || errno == EWOULDBLOCK || errno == EINTR) { @@ -449,7 +341,7 @@ static void* rtpstream_playback_thread(void* params) } /* code checked */ -static int rtpstream_start_task (rtpstream_callinfo_t *callinfo) +static int rtpstream_start_task(rtpstream_callinfo_t *callinfo) { int ready_index; int allocsize; @@ -611,10 +503,7 @@ int rtpstream_new_call (rtpstream_callinfo_t *callinfo) memset (taskinfo,0,sizeof(*taskinfo)); taskinfo->flags= TI_NULLIP; /* socket descriptors */ - taskinfo->audio_rtp_socket= -1; - taskinfo->audio_rtcp_socket= -1; - taskinfo->video_rtp_socket= -1; - taskinfo->video_rtcp_socket= -1; + taskinfo->audio_rtp_socket = -1; /* rtp stream members */ taskinfo->ssrc_id= global_ssrc_id++; /* pthread mutexes */ @@ -694,190 +583,6 @@ int rtpstream_cache_file (char *filename) return num_cached_files++; } -static int rtpstream_setsocketoptions (int sock) -{ - /* set socket non-blocking */ - int flags= fcntl(sock,F_GETFL,0); - if (fcntl(sock,F_SETFL,flags|O_NONBLOCK)==-1) { - return 0; - } - - /* set buffer size */ - unsigned int buffsize= rtp_buffsize; - - /* Increase buffer sizes for this sockets */ - if(setsockopt(sock,SOL_SOCKET,SO_SNDBUF,(char*)&buffsize,sizeof(buffsize))) { - return 0; - } - if(setsockopt(sock,SOL_SOCKET,SO_RCVBUF,(char*)&buffsize,sizeof(buffsize))) { - return 0; - } - - return 1; /* success */ -} - -/* code checked */ -static int rtpstream_get_localport (int *rtpsocket, int *rtcpsocket) -{ - int port_number; - int tries; - struct sockaddr_storage address; - - debugprint ("rtpstream_get_localport\n"); - - if (next_rtp_portsin6_addr)): - inet_pton(AF_INET,media_ip,&((_RCAST(struct sockaddr_in *,&address))->sin_addr)))!=1) { - WARNING("Could not set up media IP for RTP streaming"); - return 0; - } - - /* create new UDP listen socket */ - *rtpsocket= socket(media_ip_is_ipv6?PF_INET6:PF_INET,SOCK_DGRAM,0); - if (*rtpsocket==-1) { - WARNING("Could not open socket for RTP streaming: %s", strerror(errno)); - return 0; - } - - for (tries=0;tries(max_rtp_port-1)) { - next_rtp_port= min_rtp_port; - } - - sockaddr_update_port(&address, port_number); - if (::bind(*rtpsocket,(sockaddr *)(void *)&address, - sizeof(address)) == 0) { - break; - } - } - /* Exit here if we didn't get a suitable port for rtp stream */ - if (tries==BIND_MAX_TRIES) { - close (*rtpsocket); - *rtpsocket= -1; - WARNING("Could not bind port for RTP streaming after %d tries", tries); - return 0; - } - - if (!rtpstream_setsocketoptions (*rtpsocket)) { - close (*rtpsocket); - *rtpsocket= -1; - WARNING("Could not set socket options for RTP streaming"); - return 0; - } - - /* create socket for rtcp - ignore any errors */ - *rtcpsocket= socket(media_ip_is_ipv6?PF_INET6:PF_INET,SOCK_DGRAM,0); - if (*rtcpsocket!=-1) { - /* try to bind it to our preferred address */ - sockaddr_update_port(&address, port_number + 1); - if (::bind(*rtcpsocket,(sockaddr *)(void *)&address, - sizeof(address)) == 0) { - /* could not bind the rtcp socket to required port. so we delete it */ - close (*rtcpsocket); - *rtcpsocket= -1; - } - if (!rtpstream_setsocketoptions (*rtcpsocket)) { - close (*rtcpsocket); - *rtcpsocket= -1; - } - } - - return port_number; -} - -/* code checked */ -int rtpstream_get_audioport (rtpstream_callinfo_t *callinfo) -{ - debugprint ("rtpstream_get_audioport callinfo=%p",callinfo); - - int rtp_socket; - int rtcp_socket; - - if (!callinfo->taskinfo) { - return 0; - } - - if (callinfo->audioport) { - /* already a port assigned to this call */ - debugprint (" ==> %d\n",callinfo->audioport); - return callinfo->audioport; - } - - callinfo->audioport= rtpstream_get_localport (&rtp_socket,&rtcp_socket); - debugprint (" ==> %d\n",callinfo->audioport); - - /* assign rtp and rtcp sockets to callinfo. must assign rtcp socket first */ - callinfo->taskinfo->audio_rtcp_socket= rtcp_socket; - callinfo->taskinfo->audio_rtp_socket= rtp_socket; - - /* start playback task if not already started */ - if (!callinfo->taskinfo->parent_thread) { - if (!rtpstream_start_task (callinfo)) { - /* error starting playback task */ - return 0; - } - } - - /* make sure the new socket gets bound to destination address (if any) */ - callinfo->taskinfo->flags|= TI_RECONNECTSOCKET; - - return callinfo->audioport; -} - -/* code checked */ -int rtpstream_get_videoport (rtpstream_callinfo_t *callinfo) -{ - debugprint ("rtpstream_get_videoport callinfo=%p",callinfo); - - int rtp_socket; - int rtcp_socket; - - if (!callinfo->taskinfo) { - return 0; - } - - if (callinfo->videoport) { - /* already a port assigned to this call */ - debugprint (" ==> %d\n",callinfo->videoport); - return callinfo->videoport; - } - - callinfo->videoport= rtpstream_get_localport (&rtp_socket,&rtcp_socket); - debugprint (" ==> %d\n",callinfo->videoport); - - /* assign rtp and rtcp sockets to callinfo. must assign rtcp socket first */ - callinfo->taskinfo->video_rtcp_socket= rtcp_socket; - callinfo->taskinfo->video_rtp_socket= rtp_socket; - - /* start playback task if not already started */ - if (!callinfo->taskinfo->parent_thread) { - if (!rtpstream_start_task (callinfo)) { - /* error starting playback task */ - return 0; - } - } - - /* make sure the new socket gets bound to destination address (if any) */ - callinfo->taskinfo->flags|= TI_RECONNECTSOCKET; - - return callinfo->videoport; -} - /* code checked */ void rtpstream_set_remote (rtpstream_callinfo_t *callinfo, int ip_ver, char *ip_addr, int audio_port, int video_port) @@ -889,6 +594,8 @@ void rtpstream_set_remote (rtpstream_callinfo_t *callinfo, int ip_ver, char *ip_ unsigned count; int nonzero_ip; + /* observe that we rely on ip_ver being in sync with media_ip_is_ipv6 */ + /* we never alloc a socket here, we reuse the global media socket */ debugprint("rtpstream_set_remote callinfo=%p, ip_ver %d ip_addr %s audio %d video %d\n", callinfo, ip_ver, ip_addr, audio_port, video_port); @@ -940,38 +647,18 @@ void rtpstream_set_remote (rtpstream_callinfo_t *callinfo, int ip_ver, char *ip_ pthread_mutex_lock (&(taskinfo->mutex)); /* clear out existing addresses */ - memset (&(taskinfo->remote_audio_rtp_addr),0,sizeof(taskinfo->remote_audio_rtp_addr)); - memset (&(taskinfo->remote_audio_rtcp_addr),0,sizeof(taskinfo->remote_audio_rtcp_addr)); - memset (&(taskinfo->remote_video_rtp_addr),0,sizeof(taskinfo->remote_video_rtp_addr)); - memset (&(taskinfo->remote_video_rtcp_addr),0,sizeof(taskinfo->remote_video_rtcp_addr)); + memset(&(taskinfo->remote_audio_rtp_addr), 0, sizeof(taskinfo->remote_audio_rtp_addr)); /* Audio */ if (audio_port) { sockaddr_update_port(&address, audio_port); memcpy (&(taskinfo->remote_audio_rtp_addr),&address,sizeof(address)); - - sockaddr_update_port(&address, audio_port + 1); - memcpy (&(taskinfo->remote_audio_rtcp_addr),&address,sizeof(address)); - taskinfo->flags&= ~TI_NULL_AUDIOIP; } - /* Video */ - if (video_port) { - sockaddr_update_port(&address, video_port); - memcpy (&(taskinfo->remote_video_rtp_addr),&address,sizeof(address)); - - sockaddr_update_port(&address, video_port + 1); - memcpy (&(taskinfo->remote_video_rtcp_addr),&address,sizeof(address)); - - taskinfo->flags&= ~TI_NULL_VIDEOIP; - } - /* ok, we are done with the shared memory objects. let go mutex */ pthread_mutex_unlock (&(taskinfo->mutex)); - taskinfo->flags|= TI_RECONNECTSOCKET; - /* may want to start a playback (listen) task here if no task running? */ /* only makes sense if we decide to send 0-filled packets on idle */ } @@ -993,7 +680,15 @@ void rtpstream_play (rtpstream_callinfo_t *callinfo, rtpstream_actinfo_t *action } /* make sure we have an open socket from which to play the audio file */ - rtpstream_get_audioport (callinfo); + taskinfo->audio_rtp_socket = media_socket; + + /* start playback task if not already started */ + if (!taskinfo->parent_thread) { + if (!rtpstream_start_task(callinfo)) { + /* error starting playback task */ + return; + } + } /* save file parameter in taskinfo structure */ taskinfo->new_loop_count= actioninfo->loop_count; diff --git a/src/sipp.cpp b/src/sipp.cpp index f2ae0bde5..1be48ec44 100644 --- a/src/sipp.cpp +++ b/src/sipp.cpp @@ -257,8 +257,6 @@ struct sipp_option options_table[] = { {"mb", "Set the RTP echo buffer size (default: 2048).", SIPP_OPTION_INT, &media_bufsize, 1}, {"mp", "Set the local RTP echo port number. Default is 6000.", SIPP_OPTION_INT, &user_media_port, 1}, #ifdef RTP_STREAM - {"min_rtp_port", "Minimum port number for RTP socket range.", SIPP_OPTION_INT, &min_rtp_port, 1}, - {"max_rtp_port", "Maximum port number for RTP socket range.", SIPP_OPTION_INT, &max_rtp_port, 1}, {"rtp_payload", "RTP default payload type.", SIPP_OPTION_INT, &rtp_default_payload, 1}, {"rtp_threadtasks", "RTP number of playback tasks per thread.", SIPP_OPTION_INT, &rtp_tasks_per_thread, 1}, {"rtp_buffsize", "Set the rtp socket send/receive buffer size.", SIPP_OPTION_INT, &rtp_buffsize, 1},