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Media-aware pass-through mode


global_s
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The PBX has two ways to deal with media. The first one is that it received the media on the RTP port, decides the SRTP, decodes the RTP packets it to16-bit samples, runs it through a jitter buffer and so on just to do the same thing on the other side of the B2BUA in reverse order. This one is called "media-aware pass-through mode"

 

In many situations all that work is unnecessary. Then the PBX only needs to decode the SRTP on one side and encode it with SRTP on the other side, without looking at the content of the packet. This obviously makes the processing a lot faster, and it also deals better with network jitter. This mode is called "RTP pass-through mode". The goal of the codec negotiation is the RTP pass through mode, but for example when an operator is barging into a call or the two sides cannot or don't want to speak the same codec, there is no other choice but to use the media-aware mode.

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Thank you for the good explanation. however, having read it, I still can't figure out why in my logs the same call leg changes to Media Aware right after the other one disconnects.

 

From a user point of view, he hears no audio and thinks the call failed. In my pbx settings, lock codec durring conversation is ON and Follow RTP is ON, just for your info. No firewall involved

In the logs I see

[5] 20121119030040: Tuning to new SSRC

[5] 20121119030207: SIP Rx udp:10.246.0.112:2048:

BYE sip:50218@10.252.0.8:5060 SIP/2.0

Via: SIP/2.0/UDP 10.246.0.112:2048;branch=z9hG4bK-558bctzvotmj;rport

From: "A Group" <sip:50218@sip.ofe.com>;tag=2v6lcbp1ng

To: <sip:39022551131@si.ofe.com;phone=yes>;tag=b64cd89a88

Call-ID: 3c2dfccf780a-z8hkx7ykpdlt

CSeq: 4 BYE

Max-Forwards: 70

Contact: <sip:50218@10.246.0.112:2048;line=spr0c3mz>;reg-id=1

User-Agent: snom300/8.4.32

RTP-RxStat: Total_Rx_Pkts=4536,Rx_Pkts=0,Rx_Pkts_Lost=1,Remote_Rx_Pkts_Lost=0

RTP-TxStat: Total_Tx_Pkts=4537,Tx_Pkts=4538,Remote_Tx_Pkts=4498

Proxy-Require: buttons

Content-Length: 0

 

 

[5] 20121119030207: SIP Tx udp:10.246.0.112:2048:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 10.246.0.112:2048;branch=z9hG4bK-558bctzvotmj;rport=2048

From: "A Group" <sip:50218@sip.ofe.com>;tag=2v6lcbp1ng

To: <sip:3302454455225@sipmsk.offixspace.com;phone=yes>;tag=b64cd89a88

Call-ID: 3c2dfccf780a-z8hkx7ykpdlt

CSeq: 4 BYE

Contact: <sip:50218@10.252.0.8:5060>

User-Agent: snomONE/4.5.0.1075 Delta Aurigids

Content-Length: 0

 

 

[7] 20121119030207: 69b69727@pbx: Media-aware pass-through mode

[8] 20121119030207: call port 331: state code from 200 to 486

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The media-aware pass through is after the one call leg hung up and it actually is on the other leg. Because there is probably only one leg left, it is very understandable that there is no more RTP pass-through possible! Plug that message comes a lot later than the SSRC change.

 

I would definitively assume that the SSRC change it the problem here. The SSRC should not change unless there is a SIP-style music on hold going on or a transfer without signalling. If the SSRC in the middle of the call "out of the blue", there is something strange going on.

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The media-aware pass through is after the one call leg hung up and it actually is on the other leg. Because there is probably only one leg left, it is very understandable that there is no more RTP pass-through possible! Plug that message comes a lot later than the SSRC change.

 

I would definitively assume that the SSRC change it the problem here. The SSRC should not change unless there is a SIP-style music on hold going on or a transfer without signalling. If the SSRC in the middle of the call "out of the blue", there is something strange going on.

 

That's what I think. Do you think there is anything I could do expect a Pcap on the pbx?

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