Summary: | ASTERISK-21300: Asterisk is sending wrong codec order in the leg B of the call | ||
Reporter: | Jesus Tovar (jes_tovar) | Labels: | |
Date Opened: | 2013-03-19 02:19:20 | Date Closed: | 2013-04-09 15:31:00 |
Priority: | Critical | Regression? | |
Status: | Closed/Complete | Components: | Channels/chan_sip/CodecHandling |
Versions: | 10.12.0 11.2.0 | Frequency of Occurrence | |
Related Issues: | |||
Environment: | Centos 6.3 Linux rdns.vpdoc.com 2.6.32-279.22.1.el6.i686 #1 SMP Wed Feb 6 00:31:03 UTC 2013 i686 i686 i386 GNU/Linux | Attachments: | ( 0) SIP_Debug_PBX__V11_2_ticket.txt |
Description: | Basically the issue is focused on the codec order of the second leg of the call (PBX --> callee leg), where the PBX sets ulaw as first choice of the SDP whatever is the order in the configuration files of the peers, this order of codecs in the SDP makes the callee phone to pick up ulaw since it is already among the preference list of codecs. This issue is generating a permanent transcoding inside the server.
Whatever the codec order is in the Leg A of the call, the codec order in the Leg B is ulaw. SDP_Caller: Media Format: ITU-T G.711 PCMA Media Format: ITU-T G.711 PCMU Media Format: ITU-T G.729 SDP_Callee: Media Format: ITU-T G.711 PCMU Media Format: ITU-T G.729 Media Format: ITU-T G.711 PCMA Caller (g729/alaw/ulaw) ---> PBX --> (ulaw/alaw/g729) transcoding. Caller (alaw/ulaw/g729) ---> PBX --> (ulaw/alaw/g729) transcoding. Below there is a SIP Debug that shows clearly the failure: Trace Information: Number A (Caller): 3135 IP Address Caller: 10.1.1.237 Number B (Callee): 3088 IP Address Callee: 10.1.1.240 Sip.conf disallow=all ; First disallow all codecs allow=h264 allow=g729 ; Allow codecs in order of preference allow=alaw allow=ulaw [3135] type=peer secret=7wq canreinvite=no host=dynamic context=test callerid="PBX CLIENT t6" <3135> videosupport=yes mailbox=3135@default qualify=10000 [3088] type=peer secret=HIw canreinvite=no host=dynamic context=test callerid="PBX CLIENT t6" <3088> videosupport=yes mailbox=3088@default qualify=10000 testing*CLI> testing*CLI> sip set debug on SIP Debugging enabled <--- SIP read from UDP:10.1.1.240:32200 ---> <-------------> <--- SIP read from UDP:10.1.1.237:42936 ---> INVITE sip:3088@10.1.1.236:5066;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.1.1.237:42936;branch=z9hG4bK8156bf5778454b05 From: "VirtualHospital-RDT6" <sip:3135@10.1.1.236:5066;user=phone>;tag=2370b982d335c7c4 To: <sip:3088@10.1.1.236:5066;user=phone> Contact: <sip:3135@10.1.1.237:42936;transport=udp;user=phone> Supported: replaces, timer, path Call-ID: 87b5f796d6d01167924739f5bcb65c66@10.1.1.237 CSeq: 5661 INVITE User-Agent: Grandstream GXV3000 1.2.3.7 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 447 v=0 o=3135 8000 8000 IN IP4 10.1.1.237 s=SIP Call c=IN IP4 10.1.1.237 t=0 0 m=audio 61432 RTP/AVP 8 0 18 101 a=sendrecv a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=ptime:40 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 m=video 61434 RTP/AVP 99 a=sendrecv a=rtpmap:99 H264/90000 a=fmtp:99 profile-level-id=428014; packetization-mode=0; sprop-parameter-sets=Z0KADJWgUH5A,aM4Ecg== a=framerate:12 <-------------> --- (13 headers 18 lines) --- Sending to 10.1.1.237:42936 (no NAT) Using INVITE request as basis request - 87b5f796d6d01167924739f5bcb65c66@10.1.1.237 Found peer '3135' for '3135' from 10.1.1.237:42936 <--- Reliably Transmitting (no NAT) to 10.1.1.237:42936 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.1.1.237:42936;branch=z9hG4bK8156bf5778454b05;received=10.1.1.237 From: "VirtualHospital-RDT6" <sip:3135@10.1.1.236:5066;user=phone>;tag=2370b982d335c7c4 To: <sip:3088@10.1.1.236:5066;user=phone>;tag=as1fa61e2e Call-ID: 87b5f796d6d01167924739f5bcb65c66@10.1.1.237 CSeq: 5661 INVITE Server: PBX PBX 11.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="PBX", nonce="4b547a92" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '87b5f796d6d01167924739f5bcb65c66@10.1.1.237' in 6400 ms (Method: INVITE) <--- SIP read from UDP:10.1.1.237:42936 ---> ACK sip:3088@10.1.1.236:5066;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.1.1.237:42936;branch=z9hG4bK8156bf5778454b05 From: "VirtualHospital-RDT6" <sip:3135@10.1.1.236:5066;user=phone>;tag=2370b982d335c7c4 To: <sip:3088@10.1.1.236:5066;user=phone>;tag=as1fa61e2e Contact: <sip:3135@10.1.1.237:42936;transport=udp;user=phone> Call-ID: 87b5f796d6d01167924739f5bcb65c66@10.1.1.237 CSeq: 5661 ACK User-Agent: Grandstream GXV3000 1.2.3.7 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from UDP:10.1.1.237:42936 ---> INVITE sip:3088@10.1.1.236:5066;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.1.1.237:42936;branch=z9hG4bKec101f6561e07d30 From: "VirtualHospital-RDT6" <sip:3135@10.1.1.236:5066;user=phone>;tag=2370b982d335c7c4 To: <sip:3088@10.1.1.236:5066;user=phone> Contact: <sip:3135@10.1.1.237:42936;transport=udp;user=phone> Supported: replaces, timer, path Authorization: Digest username="3135", realm="PBX", algorithm=MD5, uri="sip:3088@10.1.1.236:5066;user=phone", nonce="4b547a92", response="199a546451f4d26bce7f8b623be1a32d" Call-ID: 87b5f796d6d01167924739f5bcb65c66@10.1.1.237 CSeq: 5662 INVITE User-Agent: Grandstream GXV3000 1.2.3.7 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 447 v=0 o=3135 8000 8001 IN IP4 10.1.1.237 s=SIP Call c=IN IP4 10.1.1.237 t=0 0 m=audio 61432 RTP/AVP 8 0 18 101 a=sendrecv a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=ptime:40 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 m=video 61434 RTP/AVP 99 a=sendrecv a=rtpmap:99 H264/90000 a=fmtp:99 profile-level-id=428014; packetization-mode=0; sprop-parameter-sets=Z0KADJWgUH5A,aM4Ecg== a=framerate:12 <-------------> --- (14 headers 18 lines) --- Sending to 10.1.1.237:42936 (no NAT) Using INVITE request as basis request - 87b5f796d6d01167924739f5bcb65c66@10.1.1.237 Found peer '3135' for '3135' from 10.1.1.237:42936 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 101 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Found RTP video format 99 Found video description format H264 for ID 99 Capabilities: us - (ulaw|alaw|g729|h264), peer - audio=(ulaw|alaw|g729)/video=(h264)/text=(nothing), combined - (ulaw|alaw|g729|h264) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 10.1.1.237:61432 Peer video RTP is at port 10.1.1.237:61434 Looking for 3088 in test (domain 10.1.1.236) list_route: hop: <sip:3135@10.1.1.237:42936;transport=udp;user=phone> <--- Transmitting (no NAT) to 10.1.1.237:42936 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.1.237:42936;branch=z9hG4bKec101f6561e07d30;received=10.1.1.237 From: "VirtualHospital-RDT6" <sip:3135@10.1.1.236:5066;user=phone>;tag=2370b982d335c7c4 To: <sip:3088@10.1.1.236:5066;user=phone> Call-ID: 87b5f796d6d01167924739f5bcb65c66@10.1.1.237 CSeq: 5662 INVITE Server: PBX PBX 11.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: <sip:3088@10.1.1.236:5066> Content-Length: 0 <------------> Audio is at 14418 Video is at 10.1.1.236:14292 Adding codec 100003 (ulaw) to SDP Adding video codec 200004 (h264) to SDP Adding codec 100008 (g729) to SDP Adding codec 100004 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.1.1.240:32200: INVITE sip:3088@10.1.1.240:32200;transport=udp;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.1.1.236:5066;branch=z9hG4bK19b27029 Max-Forwards: 70 From: "PBX CLIENT t6" <sip:3135@10.1.1.236:5066>;tag=as54d904ac To: <sip:3088@10.1.1.240:32200;transport=udp;user=phone> Contact: <sip:3135@10.1.1.236:5066> Call-ID: 20f762cf712c4a85090df4402cc71d56@10.1.1.236:5066 CSeq: 102 INVITE User-Agent: PBX PBX 11.2.1 Date: Mon, 18 Mar 2013 06:05:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 446 v=0 o=root 1112394555 1112394555 IN IP4 10.1.1.236 s=PBX PBX 11.2.1 c=IN IP4 10.1.1.236 b=CT:384 t=0 0 m=audio 14418 RTP/AVP 0 18 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=video 14292 RTP/AVP 99 a=rtpmap:99 H264/90000 a=fmtp:99 profile-level-id=428014 a=sendrecv --- <--- SIP read from UDP:10.1.1.240:32200 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.1.236:5066;branch=z9hG4bK19b27029 From: "PBX CLIENT t6" <sip:3135@10.1.1.236:5066>;tag=as54d904ac To: <sip:3088@10.1.1.240:32200;transport=udp;user=phone> Call-ID: 20f762cf712c4a85090df4402cc71d56@10.1.1.236:5066 CSeq: 102 INVITE User-Agent: Grandstream GXV3000 1.2.3.7 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from UDP:10.1.1.240:32200 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.1.1.236:5066;branch=z9hG4bK19b27029 From: "PBX CLIENT t6" <sip:3135@10.1.1.236:5066>;tag=as54d904ac To: <sip:3088@10.1.1.240:32200;transport=udp;user=phone>;tag=d137c401125656a4 Call-ID: 20f762cf712c4a85090df4402cc71d56@10.1.1.236:5066 CSeq: 102 INVITE User-Agent: Grandstream GXV3000 1.2.3.7 Contact: <sip:3088@10.1.1.240:32200;transport=udp;user=phone> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 <-------------> --- (10 headers 0 lines) --- list_route: hop: <sip:3088@10.1.1.240:32200;transport=udp;user=phone> <--- Transmitting (no NAT) to 10.1.1.237:42936 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.1.1.237:42936;branch=z9hG4bKec101f6561e07d30;received=10.1.1.237 From: "VirtualHospital-RDT6" <sip:3135@10.1.1.236:5066;user=phone>;tag=2370b982d335c7c4 To: <sip:3088@10.1.1.236:5066;user=phone>;tag=as4b2e34fd Call-ID: 87b5f796d6d01167924739f5bcb65c66@10.1.1.237 CSeq: 5662 INVITE Server: PBX PBX 11.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: <sip:3088@10.1.1.236:5066> Content-Length: 0 <------------> Really destroying SIP dialog 'dc32e644caf312f6ab938814b3914792@10.1.1.240' Method: REGISTER <--- SIP read from UDP:10.1.1.240:32200 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.236:5066;branch=z9hG4bK19b27029 From: "PBX CLIENT t6" <sip:3135@10.1.1.236:5066>;tag=as54d904ac To: <sip:3088@10.1.1.240:32200;transport=udp;user=phone>;tag=d137c401125656a4 Call-ID: 20f762cf712c4a85090df4402cc71d56@10.1.1.236:5066 CSeq: 102 INVITE User-Agent: Grandstream GXV3000 1.2.3.7 Contact: <sip:3088@10.1.1.240:32200;transport=udp;user=phone> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Supported: replaces, timer, 100rel, path Content-Length: 395 v=0 o=3088 8000 8000 IN IP4 10.1.1.240 s=SIP Call c=IN IP4 10.1.1.240 t=0 0 m=audio 3084 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 m=video 3086 RTP/AVP 99 a=sendrecv a=rtpmap:99 H264/90000 a=fmtp:99 profile-level-id=428014; packetization-mode=0; sprop-parameter-sets=Z0KADJWgUH5A,aM4Ecg== a=framerate:12 <-------------> --- (12 headers 16 lines) --- Found RTP audio format 0 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Found RTP video format 99 Found video description format H264 for ID 99 Capabilities: us - (ulaw|alaw|g729|h264), peer - audio=(ulaw)/video=(h264)/text=(nothing), combined - (ulaw|h264) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 10.1.1.240:3084 Peer video RTP is at port 10.1.1.240:3086 list_route: hop: <sip:3088@10.1.1.240:32200;transport=udp;user=phone> set_destination: Parsing <sip:3088@10.1.1.240:32200;transport=udp;user=phone> for address/port to send to set_destination: set destination to 10.1.1.240:32200 Transmitting (no NAT) to 10.1.1.240:32200: ACK sip:3088@10.1.1.240:32200;transport=udp;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.1.1.236:5066;branch=z9hG4bK45dec5cb Max-Forwards: 70 From: "PBX CLIENT t6" <sip:3135@10.1.1.236:5066>;tag=as54d904ac To: <sip:3088@10.1.1.240:32200;transport=udp;user=phone>;tag=d137c401125656a4 Contact: <sip:3135@10.1.1.236:5066> Call-ID: 20f762cf712c4a85090df4402cc71d56@10.1.1.236:5066 CSeq: 102 ACK User-Agent: PBX PBX 11.2.1 Content-Length: 0 --- Audio is at 13140 Video is at 10.1.1.236:16842 Adding video codec 200004 (h264) to SDP Adding codec 100008 (g729) to SDP Adding codec 100004 (alaw) to SDP Adding codec 100003 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 10.1.1.237:42936 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.237:42936;branch=z9hG4bKec101f6561e07d30;received=10.1.1.237 From: "VirtualHospital-RDT6" <sip:3135@10.1.1.236:5066;user=phone>;tag=2370b982d335c7c4 To: <sip:3088@10.1.1.236:5066;user=phone>;tag=as4b2e34fd Call-ID: 87b5f796d6d01167924739f5bcb65c66@10.1.1.237 CSeq: 5662 INVITE Server: PBX PBX 11.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: <sip:3088@10.1.1.236:5066> Content-Type: application/sdp Content-Length: 446 v=0 o=root 1232695348 1232695348 IN IP4 10.1.1.236 s=PBX PBX 11.2.1 c=IN IP4 10.1.1.236 b=CT:384 t=0 0 m=audio 13140 RTP/AVP 18 8 0 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=video 16842 RTP/AVP 99 a=rtpmap:99 H264/90000 a=fmtp:99 profile-level-id=428014 a=sendrecv <------------> <--- SIP read from UDP:10.1.1.237:42936 ---> ACK sip:3088@10.1.1.236:5066 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.237:42936;branch=z9hG4bKcb05b5a399021ff0 From: "VirtualHospital-RDT6" <sip:3135@10.1.1.236:5066;user=phone>;tag=2370b982d335c7c4 To: <sip:3088@10.1.1.236:5066;user=phone>;tag=as4b2e34fd Contact: <sip:3135@10.1.1.237:42936;transport=udp;user=phone> Authorization: Digest username="3135", realm="PBX", algorithm=MD5, uri="sip:3088@10.1.1.236:5066;user=phone", nonce="4b547a92", response="199a546451f4d26bce7f8b623be1a32d" Call-ID: 87b5f796d6d01167924739f5bcb65c66@10.1.1.237 CSeq: 5662 ACK User-Agent: Grandstream GXV3000 1.2.3.7 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Really destroying SIP dialog 'bf1e2303-f567eb7e-f1b433c9@10.80.7.217' Method: REGISTER <--- SIP read from UDP:10.1.1.240:32200 ---> BYE sip:3135@10.1.1.236:5066 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.240:32200;branch=z9hG4bKa26238e6e0617643 From: <sip:3088@10.1.1.240:32200;transport=udp;user=phone>;tag=d137c401125656a4 To: "PBX CLIENT t6" <sip:3135@10.1.1.236:5066>;tag=as54d904ac Call-ID: 20f762cf712c4a85090df4402cc71d56@10.1.1.236:5066 CSeq: 2839 BYE User-Agent: Grandstream GXV3000 1.2.3.7 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Reason: SIP ;text="Onhook event" Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Sending to 10.1.1.240:32200 (no NAT) Scheduling destruction of SIP dialog '20f762cf712c4a85090df4402cc71d56@10.1.1.236:5066' in 6400 ms (Method: BYE) <--- Transmitting (no NAT) to 10.1.1.240:32200 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.240:32200;branch=z9hG4bKa26238e6e0617643;received=10.1.1.240 From: <sip:3088@10.1.1.240:32200;transport=udp;user=phone>;tag=d137c401125656a4 To: "PBX CLIENT t6" <sip:3135@10.1.1.236:5066>;tag=as54d904ac Call-ID: 20f762cf712c4a85090df4402cc71d56@10.1.1.236:5066 CSeq: 2839 BYE Server: PBX PBX 11.2.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> Scheduling destruction of SIP dialog '87b5f796d6d01167924739f5bcb65c66@10.1.1.237' in 6400 ms (Method: ACK) set_destination: Parsing <sip:3135@10.1.1.237:42936;transport=udp;user=phone> for address/port to send to set_destination: set destination to 10.1.1.237:42936 Reliably Transmitting (no NAT) to 10.1.1.237:42936: BYE sip:3135@10.1.1.237:42936;transport=udp;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.1.1.236:5066;branch=z9hG4bK121773f3 Max-Forwards: 70 From: <sip:3088@10.1.1.236:5066;user=phone>;tag=as4b2e34fd To: "VirtualHospital-RDT6" <sip:3135@10.1.1.236:5066;user=phone>;tag=2370b982d335c7c4 Call-ID: 87b5f796d6d01167924739f5bcb65c66@10.1.1.237 CSeq: 102 BYE User-Agent: PBX PBX 11.2.1 Proxy-Authorization: Digest username="3135", realm="PBX", algorithm=MD5, uri="sip:10.1.1.236", nonce="4b547a92", response="d885ef09ee905ad79fb5c45e490084f1" X-PBX-HangupCause: Normal Clearing X-PBX-HangupCauseCode: 16 Content-Length: 0 --- <--- SIP read from UDP:10.1.1.237:42936 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.236:5066;branch=z9hG4bK121773f3 From: <sip:3088@10.1.1.236:5066;user=phone>;tag=as4b2e34fd To: "VirtualHospital-RDT6" <sip:3135@10.1.1.236:5066;user=phone>;tag=2370b982d335c7c4 Call-ID: 87b5f796d6d01167924739f5bcb65c66@10.1.1.237 CSeq: 102 BYE User-Agent: Grandstream GXV3000 1.2.3.7 Contact: <sip:3135@10.1.1.237:42936;transport=udp;user=phone> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Supported: replaces, timer, 100rel, path Content-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog '87b5f796d6d01167924739f5bcb65c66@10.1.1.237' Method: ACK testing*CLI> sip set debug off SIP Debugging Disabled | ||
Comments: | By: Jesus Tovar (jes_tovar) 2013-03-19 02:20:39.255-0500 Tshark and Sip Debug information By: Jesus Tovar (jes_tovar) 2013-03-19 02:23:26.224-0500 It is worth commenting this issue doesn't happened in version 1.8.X. As long as I migrated my servers to be in version 10.20 and up the problem with the wrong codec order is affecting my platform. By: Rusty Newton (rnewton) 2013-03-22 14:46:10.575-0500 Please read through https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines Please remove all debug and file configuration from the description of the issue and re-attach as *separate* files. It also looks like you included a wireshark capture in the file you did attach (which is also redundant with some of the information in the description?) Please separate out the wireshark pcap into a .pcap file and attach it to the issue. By: Rusty Newton (rnewton) 2013-04-09 15:30:46.560-0500 Reporter hasn't cleaned up the issue to follow the guidelines or provided any additional debug. Suspended due to lack of activity. Please request a bug marshal in #asterisk-bugs on the IRC network irc.freenode.net to reopen the issue should you have the additional information requested. Further information can be found at http://www.asterisk.org/developers/bug-guidelines To look into this further we would need a proper pcap attached, plus an Asterisk log (from 1.8 or 11) that includes SIP debug showing the issue along with VERBOSE and DEBUG messages enabled and turned up to level 5. |